commit d61bf803d2949fb804c61a77c53ee9dd58c6d653
author perkj <perkj@webrtc.org> Thu Mar 24 03:16:19 2016 -0700
committer Commit bot <commit-bot@chromium.org> Thu Mar 24 10:16:23 2016 +0000
tree cd7062fcd6b295303790e7855b566989607c2dfa
parent e29f0e2515934d5286950da7fc58b548132469ff

Removed MediaStreamTrackInterface::set_state
The track state should be implicitly set by the underlying source.
This removes the public method and cleans up how AudioRtpReceiver is created. Further more it cleans up how the RtpReceivers are destroyed.

Note that this cl depend on https://codereview.webrtc.org/1790633002.
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1816143002

Cr-Commit-Position: refs/heads/master@{#12115}

webrtc/api/peerconnectioninterface_unittest.cc

diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index 4d5eb96..76f9c5d 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -11,6 +11,7 @@
 #include <string>
 #include <utility>
 
+#include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/api/audiotrack.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/mediastream.h"
@@ -240,6 +241,7 @@
 
 using rtc::scoped_ptr;
 using rtc::scoped_refptr;
+using ::testing::Exactly;
 using webrtc::AudioSourceInterface;
 using webrtc::AudioTrack;
 using webrtc::AudioTrackInterface;
@@ -255,6 +257,8 @@
 using webrtc::MockDataChannelObserver;
 using webrtc::MockSetSessionDescriptionObserver;
 using webrtc::MockStatsObserver;
+using webrtc::NotifierInterface;
+using webrtc::ObserverInterface;
 using webrtc::PeerConnectionInterface;
 using webrtc::PeerConnectionObserver;
 using webrtc::RtpReceiverInterface;
@@ -387,6 +391,29 @@
   return true;
 }
 
+// Helper class to test Observer.
+class MockTrackObserver : public ObserverInterface {
+ public:
+  explicit MockTrackObserver(NotifierInterface* notifier)
+      : notifier_(notifier) {
+    notifier_->RegisterObserver(this);
+  }
+
+  ~MockTrackObserver() { Unregister(); }
+
+  void Unregister() {
+    if (notifier_) {
+      notifier_->UnregisterObserver(this);
+      notifier_ = nullptr;
+    }
+  }
+
+  MOCK_METHOD0(OnChanged, void());
+
+ private:
+  NotifierInterface* notifier_;
+};
+
 class MockPeerConnectionObserver : public PeerConnectionObserver {
  public:
   MockPeerConnectionObserver() : remote_streams_(StreamCollection::Create()) {}
@@ -1843,8 +1870,9 @@
           pc_->remote_streams()->at(0);
   EXPECT_EQ(MediaStreamTrackInterface::kEnded,
             remote_stream->GetVideoTracks()[0]->state());
-  EXPECT_EQ(MediaStreamTrackInterface::kEnded,
-            remote_stream->GetAudioTracks()[0]->state());
+  // Audio source state changes are posted.
+  EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
+                 remote_stream->GetAudioTracks()[0]->state(), 1);
 }
 
 // Test that PeerConnection methods fails gracefully after
@@ -1949,13 +1977,28 @@
   EXPECT_TRUE(DoSetRemoteDescription(desc_ms1_two_tracks.release()));
   EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
                                        reference_collection_));
+  scoped_refptr<AudioTrackInterface> audio_track2 =
+      observer_.remote_streams()->at(0)->GetAudioTracks()[1];
+  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state());
+  scoped_refptr<VideoTrackInterface> video_track2 =
+      observer_.remote_streams()->at(0)->GetVideoTracks()[1];
+  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state());
 
   // Remove the extra audio and video tracks.
   rtc::scoped_ptr<SessionDescriptionInterface> desc_ms2 =
       CreateSessionDescriptionAndReference(1, 1);
+  MockTrackObserver audio_track_observer(audio_track2);
+  MockTrackObserver video_track_observer(video_track2);
+
+  EXPECT_CALL(audio_track_observer, OnChanged()).Times(Exactly(1));
+  EXPECT_CALL(video_track_observer, OnChanged()).Times(Exactly(1));
   EXPECT_TRUE(DoSetRemoteDescription(desc_ms2.release()));
   EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
                                        reference_collection_));
+  // Audio source state changes are posted.
+  EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
+                 audio_track2->state(), 1);
+  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track2->state());
 }
 
 // This tests that remote tracks are ended if a local session description is set
@@ -2001,7 +2044,9 @@
   audio_info->rejected = true;
   EXPECT_TRUE(DoSetLocalDescription(local_offer.release()));
   EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_video->state());
-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_audio->state());
+  // Audio source state changes are posted.
+  EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
+                 remote_audio->state(), 1);
 }
 
 // This tests that we won't crash if the remote track has been removed outside