wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 1 | /* |
| 2 | * libjingle |
| 3 | * Copyright 2013, Google Inc. |
| 4 | * |
| 5 | * Redistribution and use in source and binary forms, with or without |
| 6 | * modification, are permitted provided that the following conditions are met: |
| 7 | * |
| 8 | * 1. Redistributions of source code must retain the above copyright notice, |
| 9 | * this list of conditions and the following disclaimer. |
| 10 | * 2. Redistributions in binary form must reproduce the above copyright notice, |
| 11 | * this list of conditions and the following disclaimer in the documentation |
| 12 | * and/or other materials provided with the distribution. |
| 13 | * 3. The name of the author may not be used to endorse or promote products |
| 14 | * derived from this software without specific prior written permission. |
| 15 | * |
| 16 | * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| 17 | * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| 18 | * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| 19 | * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| 20 | * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| 21 | * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| 22 | * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| 23 | * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| 24 | * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| 25 | * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| 26 | */ |
| 27 | |
| 28 | #include "talk/app/webrtc/datachannel.h" |
| 29 | #include "talk/app/webrtc/mediastreamsignaling.h" |
| 30 | #include "talk/app/webrtc/test/fakeconstraints.h" |
| 31 | #include "talk/app/webrtc/webrtcsession.h" |
| 32 | #include "talk/base/gunit.h" |
| 33 | #include "talk/media/base/fakemediaengine.h" |
| 34 | #include "talk/media/devices/fakedevicemanager.h" |
| 35 | #include "talk/session/media/channelmanager.h" |
| 36 | |
| 37 | using webrtc::MediaConstraintsInterface; |
| 38 | |
| 39 | const uint32 kFakeSsrc = 1; |
| 40 | |
| 41 | class SctpDataChannelTest : public testing::Test { |
| 42 | protected: |
| 43 | SctpDataChannelTest() |
| 44 | : media_engine_(new cricket::FakeMediaEngine), |
| 45 | data_engine_(new cricket::FakeDataEngine), |
| 46 | channel_manager_( |
| 47 | new cricket::ChannelManager(media_engine_, |
| 48 | data_engine_, |
| 49 | new cricket::FakeDeviceManager(), |
| 50 | new cricket::CaptureManager(), |
| 51 | talk_base::Thread::Current())), |
wu@webrtc.org | a054569 | 2013-08-01 22:08:14 +0000 | [diff] [blame] | 52 | ms_signaling_(new webrtc::MediaStreamSignaling( |
| 53 | talk_base::Thread::Current(), NULL)), |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 54 | session_(channel_manager_.get(), |
| 55 | talk_base::Thread::Current(), |
| 56 | talk_base::Thread::Current(), |
| 57 | NULL, |
wu@webrtc.org | a054569 | 2013-08-01 22:08:14 +0000 | [diff] [blame] | 58 | ms_signaling_.get()), |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 59 | webrtc_data_channel_(NULL) {} |
| 60 | |
| 61 | virtual void SetUp() { |
| 62 | if (!talk_base::SSLStreamAdapter::HaveDtlsSrtp()) { |
| 63 | return; |
| 64 | } |
| 65 | channel_manager_->Init(); |
| 66 | webrtc::FakeConstraints constraints; |
| 67 | constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true); |
| 68 | constraints.AddMandatory(MediaConstraintsInterface::kEnableSctpDataChannels, |
| 69 | true); |
| 70 | ASSERT_TRUE(session_.Initialize(&constraints)); |
| 71 | webrtc::SessionDescriptionInterface* offer = session_.CreateOffer(NULL); |
| 72 | ASSERT_TRUE(offer != NULL); |
| 73 | ASSERT_TRUE(session_.SetLocalDescription(offer, NULL)); |
| 74 | |
| 75 | webrtc_data_channel_ = webrtc::DataChannel::Create(&session_, "test", NULL); |
| 76 | // Connect to the media channel. |
| 77 | webrtc_data_channel_->SetSendSsrc(kFakeSsrc); |
| 78 | webrtc_data_channel_->SetReceiveSsrc(kFakeSsrc); |
| 79 | |
| 80 | session_.data_channel()->SignalReadyToSendData(true); |
| 81 | } |
| 82 | |
| 83 | void SetSendBlocked(bool blocked) { |
| 84 | bool was_blocked = data_engine_->GetChannel(0)->is_send_blocked(); |
| 85 | data_engine_->GetChannel(0)->set_send_blocked(blocked); |
| 86 | if (!blocked && was_blocked) { |
| 87 | session_.data_channel()->SignalReadyToSendData(true); |
| 88 | } |
| 89 | } |
| 90 | |
| 91 | cricket::FakeMediaEngine* media_engine_; |
| 92 | cricket::FakeDataEngine* data_engine_; |
| 93 | talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_; |
wu@webrtc.org | a054569 | 2013-08-01 22:08:14 +0000 | [diff] [blame] | 94 | talk_base::scoped_ptr<webrtc::MediaStreamSignaling> ms_signaling_; |
wu@webrtc.org | d64719d | 2013-08-01 00:00:07 +0000 | [diff] [blame] | 95 | webrtc::WebRtcSession session_; |
| 96 | talk_base::scoped_refptr<webrtc::DataChannel> webrtc_data_channel_; |
| 97 | }; |
| 98 | |
| 99 | // Tests that DataChannel::buffered_amount() is correct after the channel is |
| 100 | // blocked. |
| 101 | TEST_F(SctpDataChannelTest, BufferedAmountWhenBlocked) { |
| 102 | if (!talk_base::SSLStreamAdapter::HaveDtlsSrtp()) { |
| 103 | return; |
| 104 | } |
| 105 | webrtc::DataBuffer buffer("abcd"); |
| 106 | EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| 107 | |
| 108 | EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); |
| 109 | |
| 110 | SetSendBlocked(true); |
| 111 | const int number_of_packets = 3; |
| 112 | for (int i = 0; i < number_of_packets; ++i) { |
| 113 | EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| 114 | } |
| 115 | EXPECT_EQ(buffer.data.length() * number_of_packets, |
| 116 | webrtc_data_channel_->buffered_amount()); |
| 117 | } |
| 118 | |
| 119 | // Tests that the queued data are sent when the channel transitions from blocked |
| 120 | // to unblocked. |
| 121 | TEST_F(SctpDataChannelTest, QueuedDataSentWhenUnblocked) { |
| 122 | if (!talk_base::SSLStreamAdapter::HaveDtlsSrtp()) { |
| 123 | return; |
| 124 | } |
| 125 | webrtc::DataBuffer buffer("abcd"); |
| 126 | SetSendBlocked(true); |
| 127 | EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| 128 | |
| 129 | SetSendBlocked(false); |
| 130 | EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); |
| 131 | } |