Move talk/app/webrtc to webrtc/api

talk/app/webrtc/peerconnectionendtoend_unittest.cc

Index: talk/app/webrtc/peerconnectionendtoend_unittest.cc
diff --git a/talk/app/webrtc/peerconnectionendtoend_unittest.cc b/talk/app/webrtc/peerconnectionendtoend_unittest.cc
deleted file mode 100644
index f71ce61eb20cf966c4db61ebca5294067fdf6233..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/peerconnectionendtoend_unittest.cc
+++ /dev/null
@@ -1,387 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/app/webrtc/test/peerconnectiontestwrapper.h"
-// Notice that mockpeerconnectionobservers.h must be included after the above!
-#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
-#ifdef WEBRTC_ANDROID
-#include "talk/app/webrtc/test/androidtestinitializer.h"
-#endif
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/ssladapter.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 size_t kMaxWait = 10000;
-
-}  // namespace
-
-class PeerConnectionEndToEndTest
-    : public sigslot::has_slots<>,
-      public testing::Test {
- public:
-  typedef std::vector<rtc::scoped_refptr<DataChannelInterface> >
-      DataChannelList;
-
-  PeerConnectionEndToEndTest()
-      : caller_(new rtc::RefCountedObject<PeerConnectionTestWrapper>(
-                    "caller")),
-        callee_(new rtc::RefCountedObject<PeerConnectionTestWrapper>(
-                    "callee")) {
-#ifdef WEBRTC_ANDROID
-    webrtc::InitializeAndroidObjects();
-#endif
-  }
-
-  void CreatePcs() {
-    CreatePcs(NULL);
-  }
-
-  void CreatePcs(const MediaConstraintsInterface* pc_constraints) {
-    EXPECT_TRUE(caller_->CreatePc(pc_constraints));
-    EXPECT_TRUE(callee_->CreatePc(pc_constraints));
-    PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
-
-    caller_->SignalOnDataChannel.connect(
-        this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel);
-    callee_->SignalOnDataChannel.connect(
-        this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel);
-  }
-
-  void GetAndAddUserMedia() {
-    FakeConstraints audio_constraints;
-    FakeConstraints video_constraints;
-    GetAndAddUserMedia(true, audio_constraints, true, video_constraints);
-  }
-
-  void GetAndAddUserMedia(bool audio, FakeConstraints audio_constraints,
-                          bool video, FakeConstraints video_constraints) {
-    caller_->GetAndAddUserMedia(audio, audio_constraints,
-                                video, video_constraints);
-    callee_->GetAndAddUserMedia(audio, audio_constraints,
-                                video, video_constraints);
-  }
-
-  void Negotiate() {
-    caller_->CreateOffer(NULL);
-  }
-
-  void WaitForCallEstablished() {
-    caller_->WaitForCallEstablished();
-    callee_->WaitForCallEstablished();
-  }
-
-  void WaitForConnection() {
-    caller_->WaitForConnection();
-    callee_->WaitForConnection();
-  }
-
-  void OnCallerAddedDataChanel(DataChannelInterface* dc) {
-    caller_signaled_data_channels_.push_back(dc);
-  }
-
-  void OnCalleeAddedDataChannel(DataChannelInterface* dc) {
-    callee_signaled_data_channels_.push_back(dc);
-  }
-
-  // Tests that |dc1| and |dc2| can send to and receive from each other.
-  void TestDataChannelSendAndReceive(
-      DataChannelInterface* dc1, DataChannelInterface* dc2) {
-    rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc1_observer(
-        new webrtc::MockDataChannelObserver(dc1));
-
-    rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc2_observer(
-        new webrtc::MockDataChannelObserver(dc2));
-
-    static const std::string kDummyData = "abcdefg";
-    webrtc::DataBuffer buffer(kDummyData);
-    EXPECT_TRUE(dc1->Send(buffer));
-    EXPECT_EQ_WAIT(kDummyData, dc2_observer->last_message(), kMaxWait);
-
-    EXPECT_TRUE(dc2->Send(buffer));
-    EXPECT_EQ_WAIT(kDummyData, dc1_observer->last_message(), kMaxWait);
-
-    EXPECT_EQ(1U, dc1_observer->received_message_count());
-    EXPECT_EQ(1U, dc2_observer->received_message_count());
-  }
-
-  void WaitForDataChannelsToOpen(DataChannelInterface* local_dc,
-                                 const DataChannelList& remote_dc_list,
-                                 size_t remote_dc_index) {
-    EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait);
-
-    EXPECT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait);
-    EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
-                   remote_dc_list[remote_dc_index]->state(),
-                   kMaxWait);
-    EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id());
-  }
-
-  void CloseDataChannels(DataChannelInterface* local_dc,
-                         const DataChannelList& remote_dc_list,
-                         size_t remote_dc_index) {
-    local_dc->Close();
-    EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait);
-    EXPECT_EQ_WAIT(DataChannelInterface::kClosed,
-                   remote_dc_list[remote_dc_index]->state(),
-                   kMaxWait);
-  }
-
- protected:
-  rtc::scoped_refptr<PeerConnectionTestWrapper> caller_;
-  rtc::scoped_refptr<PeerConnectionTestWrapper> callee_;
-  DataChannelList caller_signaled_data_channels_;
-  DataChannelList callee_signaled_data_channels_;
-};
-
-// Disabled for TSan v2, see
-// https://bugs.chromium.org/p/webrtc/issues/detail?id=4719 for details.
-// Disabled for Mac, see
-// https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details.
-#if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
-TEST_F(PeerConnectionEndToEndTest, Call) {
-  CreatePcs();
-  GetAndAddUserMedia();
-  Negotiate();
-  WaitForCallEstablished();
-}
-#endif // if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
-
-TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
-  FakeConstraints pc_constraints;
-  pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
-                              false);
-  CreatePcs(&pc_constraints);
-  GetAndAddUserMedia();
-  Negotiate();
-  WaitForCallEstablished();
-}
-
-// 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.
-#if defined(MEMORY_SANITIZER)
-// Fails under MemorySanitizer:
-// See https://code.google.com/p/webrtc/issues/detail?id=3980.
-#define MAYBE_CreateDataChannelAfterNegotiate DISABLED_CreateDataChannelAfterNegotiate
-#else
-#define MAYBE_CreateDataChannelAfterNegotiate CreateDataChannelAfterNegotiate
-#endif
-TEST_F(PeerConnectionEndToEndTest, MAYBE_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();
-
-  // Creates new DataChannels after the negotiation and verifies their states.
-  rtc::scoped_refptr<DataChannelInterface> caller_dc(
-      caller_->CreateDataChannel("hello", init));
-  rtc::scoped_refptr<DataChannelInterface> callee_dc(
-      callee_->CreateDataChannel("hello", init));
-
-  WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
-  WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
-
-  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();
-  WaitForConnection();
-  WaitForDataChannelsToOpen(caller_dc_1, callee_signaled_data_channels_, 0);
-  WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1);
-
-  rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
-      new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0]));
-
-  rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
-      new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1]));
-
-  const std::string message_1 = "hello 1";
-  const std::string message_2 = "hello 2";
-
-  caller_dc_1->Send(webrtc::DataBuffer(message_1));
-  EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
-
-  caller_dc_2->Send(webrtc::DataBuffer(message_2));
-  EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
-
-  EXPECT_EQ(1U, dc_1_observer->received_message_count());
-  EXPECT_EQ(1U, dc_2_observer->received_message_count());
-}
-
-// 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.
-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);
-}