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      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/jsep.h"
     30 #include "talk/app/webrtc/mediastreamsignaling.h"
     31 #include "talk/app/webrtc/test/fakeconstraints.h"
     32 #include "talk/app/webrtc/webrtcsession.h"
     33 #include "talk/base/gunit.h"
     34 #include "talk/media/base/fakemediaengine.h"
     35 #include "talk/media/devices/fakedevicemanager.h"
     36 #include "talk/session/media/channelmanager.h"
     37 
     38 using webrtc::CreateSessionDescriptionObserver;
     39 using webrtc::MediaConstraintsInterface;
     40 using webrtc::SessionDescriptionInterface;
     41 
     42 const uint32 kFakeSsrc = 1;
     43 
     44 class CreateSessionDescriptionObserverForTest
     45     : public talk_base::RefCountedObject<CreateSessionDescriptionObserver> {
     46  public:
     47   virtual void OnSuccess(SessionDescriptionInterface* desc) {
     48     description_.reset(desc);
     49   }
     50   virtual void OnFailure(const std::string& error) {}
     51 
     52   SessionDescriptionInterface* description() { return description_.get(); }
     53 
     54   SessionDescriptionInterface* ReleaseDescription() {
     55     return description_.release();
     56   }
     57 
     58  protected:
     59   ~CreateSessionDescriptionObserverForTest() {}
     60 
     61  private:
     62   talk_base::scoped_ptr<SessionDescriptionInterface> description_;
     63 };
     64 
     65 class SctpDataChannelTest : public testing::Test {
     66  protected:
     67   SctpDataChannelTest()
     68       : media_engine_(new cricket::FakeMediaEngine),
     69         data_engine_(new cricket::FakeDataEngine),
     70         channel_manager_(
     71             new cricket::ChannelManager(media_engine_,
     72                                         data_engine_,
     73                                         new cricket::FakeDeviceManager(),
     74                                         new cricket::CaptureManager(),
     75                                         talk_base::Thread::Current())),
     76         media_stream_signaling_(
     77             new webrtc::MediaStreamSignaling(talk_base::Thread::Current(),
     78                                              NULL)),
     79         session_(channel_manager_.get(),
     80                  talk_base::Thread::Current(),
     81                  talk_base::Thread::Current(),
     82                  NULL,
     83                  media_stream_signaling_.get()),
     84         webrtc_data_channel_(NULL) {}
     85 
     86   virtual void SetUp() {
     87     if (!talk_base::SSLStreamAdapter::HaveDtlsSrtp()) {
     88       return;
     89     }
     90     channel_manager_->Init();
     91     webrtc::FakeConstraints constraints;
     92     constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true);
     93     constraints.AddMandatory(MediaConstraintsInterface::kEnableSctpDataChannels,
     94                              true);
     95     ASSERT_TRUE(session_.Initialize(&constraints, NULL));
     96     talk_base::scoped_refptr<CreateSessionDescriptionObserverForTest> observer
     97         = new CreateSessionDescriptionObserverForTest();
     98     session_.CreateOffer(observer.get(), NULL);
     99     EXPECT_TRUE_WAIT(observer->description() != NULL, 1000);
    100     ASSERT_TRUE(observer->description() != NULL);
    101     ASSERT_TRUE(session_.SetLocalDescription(observer->ReleaseDescription(),
    102                                              NULL));
    103 
    104     webrtc_data_channel_ = webrtc::DataChannel::Create(&session_, "test", NULL);
    105     // Connect to the media channel.
    106     webrtc_data_channel_->SetSendSsrc(kFakeSsrc);
    107     webrtc_data_channel_->SetReceiveSsrc(kFakeSsrc);
    108 
    109     session_.data_channel()->SignalReadyToSendData(true);
    110   }
    111 
    112   void SetSendBlocked(bool blocked) {
    113     bool was_blocked = data_engine_->GetChannel(0)->is_send_blocked();
    114     data_engine_->GetChannel(0)->set_send_blocked(blocked);
    115     if (!blocked && was_blocked) {
    116       session_.data_channel()->SignalReadyToSendData(true);
    117     }
    118   }
    119 
    120   cricket::FakeMediaEngine* media_engine_;
    121   cricket::FakeDataEngine* data_engine_;
    122   talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_;
    123   talk_base::scoped_ptr<webrtc::MediaStreamSignaling> media_stream_signaling_;
    124   webrtc::WebRtcSession session_;
    125   talk_base::scoped_refptr<webrtc::DataChannel> webrtc_data_channel_;
    126 };
    127 
    128 // Tests that DataChannel::buffered_amount() is correct after the channel is
    129 // blocked.
    130 TEST_F(SctpDataChannelTest, BufferedAmountWhenBlocked) {
    131   if (!talk_base::SSLStreamAdapter::HaveDtlsSrtp()) {
    132     return;
    133   }
    134   webrtc::DataBuffer buffer("abcd");
    135   EXPECT_TRUE(webrtc_data_channel_->Send(buffer));
    136 
    137   EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount());
    138 
    139   SetSendBlocked(true);
    140   const int number_of_packets = 3;
    141   for (int i = 0; i < number_of_packets; ++i) {
    142     EXPECT_TRUE(webrtc_data_channel_->Send(buffer));
    143   }
    144   EXPECT_EQ(buffer.data.length() * number_of_packets,
    145             webrtc_data_channel_->buffered_amount());
    146 }
    147 
    148 // Tests that the queued data are sent when the channel transitions from blocked
    149 // to unblocked.
    150 TEST_F(SctpDataChannelTest, QueuedDataSentWhenUnblocked) {
    151   if (!talk_base::SSLStreamAdapter::HaveDtlsSrtp()) {
    152     return;
    153   }
    154   webrtc::DataBuffer buffer("abcd");
    155   SetSendBlocked(true);
    156   EXPECT_TRUE(webrtc_data_channel_->Send(buffer));
    157 
    158   SetSendBlocked(false);
    159   EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount());
    160 }
    161