Chromium Code Reviews| Index: webrtc/pc/peerconnectionendtoend_datachannel_only_unittest.cc |
| diff --git a/webrtc/pc/peerconnectionendtoend_datachannel_only_unittest.cc b/webrtc/pc/peerconnectionendtoend_datachannel_only_unittest.cc |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..2f7eae109059d9df083e71d848740e3911993302 |
| --- /dev/null |
| +++ b/webrtc/pc/peerconnectionendtoend_datachannel_only_unittest.cc |
| @@ -0,0 +1,187 @@ |
| +/* |
| + * 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/base/gunit.h" |
| +#include "webrtc/base/logging.h" |
| +#include "webrtc/base/ptr_util.h" |
| +#include "webrtc/base/ssladapter.h" |
| +#include "webrtc/base/sslstreamadapter.h" |
| +#include "webrtc/base/stringencode.h" |
| +#include "webrtc/base/stringutils.h" |
| +#include "webrtc/base/thread.h" |
| +#ifdef WEBRTC_ANDROID |
| +#include "webrtc/pc/test/androidtestinitializer.h" |
| +#endif |
| +#include "webrtc/pc/test/peerconnectiontestwrapper.h" |
| +// Notice that mockpeerconnectionobservers.h must be included after the above! |
| +#include "webrtc/pc/test/mockpeerconnectionobservers.h" |
| + |
| +using webrtc::DataChannelInterface; |
| +using webrtc::FakeConstraints; |
| +using webrtc::MediaConstraintsInterface; |
| +using webrtc::MediaStreamInterface; |
| +using webrtc::PeerConnectionInterface; |
| + |
| +namespace { |
| + |
| +const int kMaxWait = 10000; |
| + |
| +} // namespace |
| + |
| +class PeerConnectionEndToEndTest : public sigslot::has_slots<>, |
| + public testing::Test { |
| + public: |
| + typedef std::vector<rtc::scoped_refptr<DataChannelInterface> > |
| + DataChannelList; |
| + |
| + PeerConnectionEndToEndTest() { |
| + RTC_CHECK(network_thread_.Start()); |
| + RTC_CHECK(worker_thread_.Start()); |
| + caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>( |
| + "caller", &network_thread_, &worker_thread_); |
| + callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>( |
| + "callee", &network_thread_, &worker_thread_); |
| + webrtc::PeerConnectionInterface::IceServer ice_server; |
| + ice_server.uri = "stun:stun.l.google.com:19302"; |
| + config_.servers.push_back(ice_server); |
| + |
| +#ifdef WEBRTC_ANDROID |
| + webrtc::InitializeAndroidObjects(); |
| +#endif |
| + } |
| + |
| + void CreatePcs( |
| + const MediaConstraintsInterface* pc_constraints, |
| + rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, |
| + rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) { |
| + EXPECT_TRUE(caller_->CreatePc( |
| + pc_constraints, config_, audio_encoder_factory, audio_decoder_factory)); |
| + EXPECT_TRUE(callee_->CreatePc( |
| + pc_constraints, config_, audio_encoder_factory, audio_decoder_factory)); |
| + PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get()); |
| + |
| + caller_->SignalOnDataChannel.connect( |
| + this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel); |
| + callee_->SignalOnDataChannel.connect( |
| + this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel); |
| + } |
| + |
| + 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) { |
| + std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer( |
| + new webrtc::MockDataChannelObserver(dc1)); |
| + |
| + std::unique_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::Thread network_thread_; |
| + rtc::Thread worker_thread_; |
| + rtc::scoped_refptr<PeerConnectionTestWrapper> caller_; |
| + rtc::scoped_refptr<PeerConnectionTestWrapper> callee_; |
| + DataChannelList caller_signaled_data_channels_; |
| + DataChannelList callee_signaled_data_channels_; |
| + webrtc::PeerConnectionInterface::RTCConfiguration config_; |
| +}; |
| + |
| +#ifdef HAVE_SCTP |
|
Taylor Brandstetter
2017/05/11 04:43:11
I was wondering what would happen if you built pee
Zhi Huang
2017/05/12 20:05:33
Good question!
I changed the place of the #ifdef.
|
| +// Verifies that the message is received by the right remote DataChannel when |
| +// there are multiple DataChannels. |
| +TEST_F(PeerConnectionEndToEndTest, |
| + MessageTransferBetweenTwoPairsOfDataChannels) { |
| + CreatePcs(nullptr, rtc::scoped_refptr<webrtc::AudioEncoderFactory>(), |
| + rtc::scoped_refptr<webrtc::AudioDecoderFactory>()); |
| + |
| + 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); |
| + |
| + std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer( |
| + new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0])); |
| + |
| + std::unique_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()); |
| +} |
| +#endif // HAVE_SCTP |