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| 1 /* |
| 2 * Copyright 2013 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #include <memory> |
| 12 |
| 13 #include "webrtc/base/gunit.h" |
| 14 #include "webrtc/base/logging.h" |
| 15 #include "webrtc/base/ptr_util.h" |
| 16 #include "webrtc/base/ssladapter.h" |
| 17 #include "webrtc/base/sslstreamadapter.h" |
| 18 #include "webrtc/base/stringencode.h" |
| 19 #include "webrtc/base/stringutils.h" |
| 20 #include "webrtc/base/thread.h" |
| 21 #ifdef WEBRTC_ANDROID |
| 22 #include "webrtc/pc/test/androidtestinitializer.h" |
| 23 #endif |
| 24 #include "webrtc/pc/test/peerconnectiontestwrapper.h" |
| 25 // Notice that mockpeerconnectionobservers.h must be included after the above! |
| 26 #include "webrtc/pc/test/mockpeerconnectionobservers.h" |
| 27 |
| 28 using webrtc::DataChannelInterface; |
| 29 using webrtc::FakeConstraints; |
| 30 using webrtc::MediaConstraintsInterface; |
| 31 using webrtc::MediaStreamInterface; |
| 32 using webrtc::PeerConnectionInterface; |
| 33 |
| 34 namespace { |
| 35 |
| 36 const int kMaxWait = 10000; |
| 37 |
| 38 } // namespace |
| 39 |
| 40 class PeerConnectionEndToEndTest : public sigslot::has_slots<>, |
| 41 public testing::Test { |
| 42 public: |
| 43 typedef std::vector<rtc::scoped_refptr<DataChannelInterface> > |
| 44 DataChannelList; |
| 45 |
| 46 PeerConnectionEndToEndTest() { |
| 47 RTC_CHECK(network_thread_.Start()); |
| 48 RTC_CHECK(worker_thread_.Start()); |
| 49 caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>( |
| 50 "caller", &network_thread_, &worker_thread_); |
| 51 callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>( |
| 52 "callee", &network_thread_, &worker_thread_); |
| 53 webrtc::PeerConnectionInterface::IceServer ice_server; |
| 54 ice_server.uri = "stun:stun.l.google.com:19302"; |
| 55 config_.servers.push_back(ice_server); |
| 56 |
| 57 #ifdef WEBRTC_ANDROID |
| 58 webrtc::InitializeAndroidObjects(); |
| 59 #endif |
| 60 } |
| 61 |
| 62 void CreatePcs( |
| 63 const MediaConstraintsInterface* pc_constraints, |
| 64 rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, |
| 65 rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) { |
| 66 EXPECT_TRUE(caller_->CreatePc( |
| 67 pc_constraints, config_, audio_encoder_factory, audio_decoder_factory)); |
| 68 EXPECT_TRUE(callee_->CreatePc( |
| 69 pc_constraints, config_, audio_encoder_factory, audio_decoder_factory)); |
| 70 PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get()); |
| 71 |
| 72 caller_->SignalOnDataChannel.connect( |
| 73 this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel); |
| 74 callee_->SignalOnDataChannel.connect( |
| 75 this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel); |
| 76 } |
| 77 |
| 78 void Negotiate() { caller_->CreateOffer(NULL); } |
| 79 |
| 80 void WaitForCallEstablished() { |
| 81 caller_->WaitForCallEstablished(); |
| 82 callee_->WaitForCallEstablished(); |
| 83 } |
| 84 |
| 85 void WaitForConnection() { |
| 86 caller_->WaitForConnection(); |
| 87 callee_->WaitForConnection(); |
| 88 } |
| 89 |
| 90 void OnCallerAddedDataChanel(DataChannelInterface* dc) { |
| 91 caller_signaled_data_channels_.push_back(dc); |
| 92 } |
| 93 |
| 94 void OnCalleeAddedDataChannel(DataChannelInterface* dc) { |
| 95 callee_signaled_data_channels_.push_back(dc); |
| 96 } |
| 97 |
| 98 // Tests that |dc1| and |dc2| can send to and receive from each other. |
| 99 void TestDataChannelSendAndReceive(DataChannelInterface* dc1, |
| 100 DataChannelInterface* dc2) { |
| 101 std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer( |
| 102 new webrtc::MockDataChannelObserver(dc1)); |
| 103 |
| 104 std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer( |
| 105 new webrtc::MockDataChannelObserver(dc2)); |
| 106 |
| 107 static const std::string kDummyData = "abcdefg"; |
| 108 webrtc::DataBuffer buffer(kDummyData); |
| 109 EXPECT_TRUE(dc1->Send(buffer)); |
| 110 EXPECT_EQ_WAIT(kDummyData, dc2_observer->last_message(), kMaxWait); |
| 111 |
| 112 EXPECT_TRUE(dc2->Send(buffer)); |
| 113 EXPECT_EQ_WAIT(kDummyData, dc1_observer->last_message(), kMaxWait); |
| 114 |
| 115 EXPECT_EQ(1U, dc1_observer->received_message_count()); |
| 116 EXPECT_EQ(1U, dc2_observer->received_message_count()); |
| 117 } |
| 118 |
| 119 void WaitForDataChannelsToOpen(DataChannelInterface* local_dc, |
| 120 const DataChannelList& remote_dc_list, |
| 121 size_t remote_dc_index) { |
| 122 EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait); |
| 123 |
| 124 EXPECT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait); |
| 125 EXPECT_EQ_WAIT(DataChannelInterface::kOpen, |
| 126 remote_dc_list[remote_dc_index]->state(), kMaxWait); |
| 127 EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id()); |
| 128 } |
| 129 |
| 130 void CloseDataChannels(DataChannelInterface* local_dc, |
| 131 const DataChannelList& remote_dc_list, |
| 132 size_t remote_dc_index) { |
| 133 local_dc->Close(); |
| 134 EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait); |
| 135 EXPECT_EQ_WAIT(DataChannelInterface::kClosed, |
| 136 remote_dc_list[remote_dc_index]->state(), kMaxWait); |
| 137 } |
| 138 |
| 139 protected: |
| 140 rtc::Thread network_thread_; |
| 141 rtc::Thread worker_thread_; |
| 142 rtc::scoped_refptr<PeerConnectionTestWrapper> caller_; |
| 143 rtc::scoped_refptr<PeerConnectionTestWrapper> callee_; |
| 144 DataChannelList caller_signaled_data_channels_; |
| 145 DataChannelList callee_signaled_data_channels_; |
| 146 webrtc::PeerConnectionInterface::RTCConfiguration config_; |
| 147 }; |
| 148 |
| 149 // Verifies that the message is received by the right remote DataChannel. |
| 150 TEST_F(PeerConnectionEndToEndTest, |
| 151 MessageTransferBetweenTwoPairsOfDataChannels) { |
| 152 #ifdef HAVE_SCTP |
| 153 CreatePcs(nullptr, rtc::scoped_refptr<webrtc::AudioEncoderFactory>(), |
| 154 rtc::scoped_refptr<webrtc::AudioDecoderFactory>()); |
| 155 |
| 156 webrtc::DataChannelInit init; |
| 157 |
| 158 rtc::scoped_refptr<DataChannelInterface> caller_dc_1( |
| 159 caller_->CreateDataChannel("data", init)); |
| 160 rtc::scoped_refptr<DataChannelInterface> caller_dc_2( |
| 161 caller_->CreateDataChannel("data", init)); |
| 162 |
| 163 Negotiate(); |
| 164 WaitForConnection(); |
| 165 WaitForDataChannelsToOpen(caller_dc_1, callee_signaled_data_channels_, 0); |
| 166 WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1); |
| 167 |
| 168 std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer( |
| 169 new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0])); |
| 170 |
| 171 std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer( |
| 172 new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1])); |
| 173 |
| 174 const std::string message_1 = "hello 1"; |
| 175 const std::string message_2 = "hello 2"; |
| 176 |
| 177 caller_dc_1->Send(webrtc::DataBuffer(message_1)); |
| 178 EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait); |
| 179 |
| 180 caller_dc_2->Send(webrtc::DataBuffer(message_2)); |
| 181 EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait); |
| 182 |
| 183 EXPECT_EQ(1U, dc_1_observer->received_message_count()); |
| 184 EXPECT_EQ(1U, dc_2_observer->received_message_count()); |
| 185 #endif // HAVE_SCTP |
| 186 } |
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