Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/mediastream_unittest.cc b/talk/app/webrtc/mediastream_unittest.cc
new file mode 100644
index 0000000..bb2d50e
--- /dev/null
+++ b/talk/app/webrtc/mediastream_unittest.cc
@@ -0,0 +1,162 @@
+/*
+ * libjingle
+ * Copyright 2011, 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 <string>
+
+#include "talk/app/webrtc/audiotrack.h"
+#include "talk/app/webrtc/mediastream.h"
+#include "talk/app/webrtc/videotrack.h"
+#include "talk/base/refcount.h"
+#include "talk/base/scoped_ptr.h"
+#include "talk/base/gunit.h"
+#include "testing/base/public/gmock.h"
+
+static const char kStreamLabel1[] = "local_stream_1";
+static const char kVideoTrackId[] = "dummy_video_cam_1";
+static const char kAudioTrackId[] = "dummy_microphone_1";
+
+using talk_base::scoped_refptr;
+using ::testing::Exactly;
+
+namespace webrtc {
+
+// Helper class to test Observer.
+class MockObserver : public ObserverInterface {
+ public:
+ MockObserver() {}
+
+ MOCK_METHOD0(OnChanged, void());
+};
+
+class MediaStreamTest: public testing::Test {
+ protected:
+ virtual void SetUp() {
+ stream_ = MediaStream::Create(kStreamLabel1);
+ ASSERT_TRUE(stream_.get() != NULL);
+
+ video_track_ = VideoTrack::Create(kVideoTrackId, NULL);
+ ASSERT_TRUE(video_track_.get() != NULL);
+ EXPECT_EQ(MediaStreamTrackInterface::kInitializing, video_track_->state());
+
+ audio_track_ = AudioTrack::Create(kAudioTrackId, NULL);
+
+ ASSERT_TRUE(audio_track_.get() != NULL);
+ EXPECT_EQ(MediaStreamTrackInterface::kInitializing, audio_track_->state());
+
+ EXPECT_TRUE(stream_->AddTrack(video_track_));
+ EXPECT_FALSE(stream_->AddTrack(video_track_));
+ EXPECT_TRUE(stream_->AddTrack(audio_track_));
+ EXPECT_FALSE(stream_->AddTrack(audio_track_));
+ }
+
+ void ChangeTrack(MediaStreamTrackInterface* track) {
+ MockObserver observer;
+ track->RegisterObserver(&observer);
+
+ EXPECT_CALL(observer, OnChanged())
+ .Times(Exactly(1));
+ track->set_enabled(false);
+ EXPECT_FALSE(track->enabled());
+
+ EXPECT_CALL(observer, OnChanged())
+ .Times(Exactly(1));
+ track->set_state(MediaStreamTrackInterface::kLive);
+ EXPECT_EQ(MediaStreamTrackInterface::kLive, track->state());
+ }
+
+ scoped_refptr<MediaStreamInterface> stream_;
+ scoped_refptr<AudioTrackInterface> audio_track_;
+ scoped_refptr<VideoTrackInterface> video_track_;
+};
+
+TEST_F(MediaStreamTest, GetTrackInfo) {
+ ASSERT_EQ(1u, stream_->GetVideoTracks().size());
+ ASSERT_EQ(1u, stream_->GetAudioTracks().size());
+
+ // Verify the video track.
+ scoped_refptr<webrtc::MediaStreamTrackInterface> video_track(
+ stream_->GetVideoTracks()[0]);
+ EXPECT_EQ(0, video_track->id().compare(kVideoTrackId));
+ EXPECT_TRUE(video_track->enabled());
+
+ ASSERT_EQ(1u, stream_->GetVideoTracks().size());
+ EXPECT_TRUE(stream_->GetVideoTracks()[0].get() == video_track.get());
+ EXPECT_TRUE(stream_->FindVideoTrack(video_track->id()).get()
+ == video_track.get());
+ video_track = stream_->GetVideoTracks()[0];
+ EXPECT_EQ(0, video_track->id().compare(kVideoTrackId));
+ EXPECT_TRUE(video_track->enabled());
+
+ // Verify the audio track.
+ scoped_refptr<webrtc::MediaStreamTrackInterface> audio_track(
+ stream_->GetAudioTracks()[0]);
+ EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId));
+ EXPECT_TRUE(audio_track->enabled());
+ ASSERT_EQ(1u, stream_->GetAudioTracks().size());
+ EXPECT_TRUE(stream_->GetAudioTracks()[0].get() == audio_track.get());
+ EXPECT_TRUE(stream_->FindAudioTrack(audio_track->id()).get()
+ == audio_track.get());
+ audio_track = stream_->GetAudioTracks()[0];
+ EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId));
+ EXPECT_TRUE(audio_track->enabled());
+}
+
+TEST_F(MediaStreamTest, RemoveTrack) {
+ MockObserver observer;
+ stream_->RegisterObserver(&observer);
+
+ EXPECT_CALL(observer, OnChanged())
+ .Times(Exactly(2));
+
+ EXPECT_TRUE(stream_->RemoveTrack(audio_track_));
+ EXPECT_FALSE(stream_->RemoveTrack(audio_track_));
+ EXPECT_EQ(0u, stream_->GetAudioTracks().size());
+ EXPECT_EQ(0u, stream_->GetAudioTracks().size());
+
+ EXPECT_TRUE(stream_->RemoveTrack(video_track_));
+ EXPECT_FALSE(stream_->RemoveTrack(video_track_));
+
+ EXPECT_EQ(0u, stream_->GetVideoTracks().size());
+ EXPECT_EQ(0u, stream_->GetVideoTracks().size());
+
+ EXPECT_FALSE(stream_->RemoveTrack(static_cast<AudioTrackInterface*>(NULL)));
+ EXPECT_FALSE(stream_->RemoveTrack(static_cast<VideoTrackInterface*>(NULL)));
+}
+
+TEST_F(MediaStreamTest, ChangeVideoTrack) {
+ scoped_refptr<webrtc::VideoTrackInterface> video_track(
+ stream_->GetVideoTracks()[0]);
+ ChangeTrack(video_track.get());
+}
+
+TEST_F(MediaStreamTest, ChangeAudioTrack) {
+ scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ stream_->GetAudioTracks()[0]);
+ ChangeTrack(audio_track.get());
+}
+
+} // namespace webrtc
