Update libjingle to 50191337.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1885005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4461 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/datachannel_unittest.cc b/talk/app/webrtc/datachannel_unittest.cc
new file mode 100644
index 0000000..d3faf17
--- /dev/null
+++ b/talk/app/webrtc/datachannel_unittest.cc
@@ -0,0 +1,129 @@
+/*
+ * 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/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::MediaConstraintsInterface;
+
+const uint32 kFakeSsrc = 1;
+
+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())),
+ session_(channel_manager_.get(),
+ talk_base::Thread::Current(),
+ talk_base::Thread::Current(),
+ NULL,
+ new webrtc::MediaStreamSignaling(talk_base::Thread::Current(),
+ NULL)),
+ 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));
+ webrtc::SessionDescriptionInterface* offer = session_.CreateOffer(NULL);
+ ASSERT_TRUE(offer != NULL);
+ ASSERT_TRUE(session_.SetLocalDescription(offer, 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_;
+ 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());
+}