Switch LiveAudio to source provider solution.

content/renderer/media/webrtc_local_audio_track_unittest.cc

Index: content/renderer/media/webrtc_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc
index 7d125dcdf825e226b95ab1a842f5763ddce6b101..316218cb64e4d694d9c29d91b7586c0102e4422f 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -134,6 +134,7 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     capturer_ = WebRtcAudioCapturer::CreateCapturer();
+    capturer_->EnablePeerConnectionMode();

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

why do we need this?  Should we run all tests twice, once with and once without
peerconnection mode enabled?  (it's possible to parameterize tests like that)

[no longer working on chromium, 2013/09/10 12:43:15]

Oh, we don't need this. Removed.

     capturer_source_ = new MockCapturerSource();
     EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0))
         .WillOnce(Return());
@@ -167,7 +168,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
   EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   track->Start();
   EXPECT_TRUE(track->enabled());
@@ -213,7 +214,7 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
   EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
-    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
   track->Start();
   static_cast<webrtc::AudioTrackInterface*>(track.get())->
@@ -263,7 +264,7 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
   EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
-    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
   track_1->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
@@ -288,7 +289,7 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
-    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
@@ -343,7 +344,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   track->Start();
 
@@ -362,7 +363,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
   EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(SignalEvent(&event));
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
@@ -382,7 +383,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
   // since it has been started.
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
@@ -415,7 +416,7 @@ TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   track->Start();
 
@@ -442,7 +443,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   track_1->Start();
 
@@ -482,7 +483,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
   // Setup the second audio track, connect it to the new capturer and start it.
   EXPECT_CALL(*new_source.get(), Start()).Times(1);
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
-      WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL, NULL,
           &constraints);
   track_2->Start();