| /* |
| * 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/datachannel.h" |
| #include "talk/app/webrtc/jsep.h" |
| #include "talk/app/webrtc/mediastreamsignaling.h" |
| #include "talk/app/webrtc/test/fakeconstraints.h" |
| #include "talk/app/webrtc/webrtcsession.h" |
| #include "talk/base/gunit.h" |
| #include "talk/media/base/fakemediaengine.h" |
| #include "talk/media/devices/fakedevicemanager.h" |
| #include "talk/session/media/channelmanager.h" |
| |
| using webrtc::CreateSessionDescriptionObserver; |
| using webrtc::MediaConstraintsInterface; |
| using webrtc::SessionDescriptionInterface; |
| |
| const uint32 kFakeSsrc = 1; |
| |
| class CreateSessionDescriptionObserverForTest |
| : public talk_base::RefCountedObject<CreateSessionDescriptionObserver> { |
| public: |
| CreateSessionDescriptionObserverForTest() : description_(NULL) {} |
| |
| virtual void OnSuccess(SessionDescriptionInterface* desc) { |
| description_ = desc; |
| } |
| virtual void OnFailure(const std::string& error) {} |
| |
| SessionDescriptionInterface* description() { return description_; } |
| |
| protected: |
| ~CreateSessionDescriptionObserverForTest() {} |
| |
| private: |
| SessionDescriptionInterface* description_; |
| }; |
| |
| class SctpDataChannelTest : public testing::Test { |
| protected: |
| SctpDataChannelTest() |
| : media_engine_(new cricket::FakeMediaEngine), |
| data_engine_(new cricket::FakeDataEngine), |
| channel_manager_( |
| new cricket::ChannelManager(media_engine_, |
| data_engine_, |
| new cricket::FakeDeviceManager(), |
| new cricket::CaptureManager(), |
| talk_base::Thread::Current())), |
| media_stream_signaling_( |
| new webrtc::MediaStreamSignaling(talk_base::Thread::Current(), |
| NULL)), |
| session_(channel_manager_.get(), |
| talk_base::Thread::Current(), |
| talk_base::Thread::Current(), |
| NULL, |
| media_stream_signaling_.get()), |
| webrtc_data_channel_(NULL) {} |
| |
| virtual void SetUp() { |
| if (!talk_base::SSLStreamAdapter::HaveDtlsSrtp()) { |
| return; |
| } |
| channel_manager_->Init(); |
| webrtc::FakeConstraints constraints; |
| constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true); |
| constraints.AddMandatory(MediaConstraintsInterface::kEnableSctpDataChannels, |
| true); |
| ASSERT_TRUE(session_.Initialize(&constraints, NULL)); |
| talk_base::scoped_refptr<CreateSessionDescriptionObserverForTest> observer |
| = new CreateSessionDescriptionObserverForTest(); |
| session_.CreateOffer(observer.get(), NULL); |
| EXPECT_TRUE_WAIT(observer->description() != NULL, 1000); |
| ASSERT_TRUE(observer->description() != NULL); |
| ASSERT_TRUE(session_.SetLocalDescription(observer->description(), NULL)); |
| |
| webrtc_data_channel_ = webrtc::DataChannel::Create(&session_, "test", NULL); |
| // Connect to the media channel. |
| webrtc_data_channel_->SetSendSsrc(kFakeSsrc); |
| webrtc_data_channel_->SetReceiveSsrc(kFakeSsrc); |
| |
| session_.data_channel()->SignalReadyToSendData(true); |
| } |
| |
| void SetSendBlocked(bool blocked) { |
| bool was_blocked = data_engine_->GetChannel(0)->is_send_blocked(); |
| data_engine_->GetChannel(0)->set_send_blocked(blocked); |
| if (!blocked && was_blocked) { |
| session_.data_channel()->SignalReadyToSendData(true); |
| } |
| } |
| |
| cricket::FakeMediaEngine* media_engine_; |
| cricket::FakeDataEngine* data_engine_; |
| talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_; |
| talk_base::scoped_ptr<webrtc::MediaStreamSignaling> media_stream_signaling_; |
| webrtc::WebRtcSession session_; |
| talk_base::scoped_refptr<webrtc::DataChannel> webrtc_data_channel_; |
| }; |
| |
| // Tests that DataChannel::buffered_amount() is correct after the channel is |
| // blocked. |
| TEST_F(SctpDataChannelTest, BufferedAmountWhenBlocked) { |
| if (!talk_base::SSLStreamAdapter::HaveDtlsSrtp()) { |
| return; |
| } |
| webrtc::DataBuffer buffer("abcd"); |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| |
| EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); |
| |
| SetSendBlocked(true); |
| const int number_of_packets = 3; |
| for (int i = 0; i < number_of_packets; ++i) { |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| } |
| EXPECT_EQ(buffer.data.length() * number_of_packets, |
| webrtc_data_channel_->buffered_amount()); |
| } |
| |
| // Tests that the queued data are sent when the channel transitions from blocked |
| // to unblocked. |
| TEST_F(SctpDataChannelTest, QueuedDataSentWhenUnblocked) { |
| if (!talk_base::SSLStreamAdapter::HaveDtlsSrtp()) { |
| return; |
| } |
| webrtc::DataBuffer buffer("abcd"); |
| SetSendBlocked(true); |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| |
| SetSendBlocked(false); |
| EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); |
| } |