Update talk to 60923971

Review URL: https://webrtc-codereview.appspot.com/7909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d

8 files changed, 115 insertions, 15 deletions

diff --git a/talk/app/webrtc/jsepsessiondescription.cc b/talk/app/webrtc/jsepsessiondescription.cc
index 8ec145808a..d7b37b5d76 100644
--- a/talk/app/webrtc/jsepsessiondescription.cc
+++ b/talk/app/webrtc/jsepsessiondescription.cc
@@ -118,9 +118,6 @@ bool JsepSessionDescription::AddCandidate(
   }
   if (mediasection_index >= number_of_mediasections())
     return false;
-  if (candidate_collection_[mediasection_index].HasCandidate(candidate)) {
-    return true;  // Silently ignore this candidate if we already have it.
-  }
   const std::string content_name =
       description_->contents()[mediasection_index].name;
   const cricket::TransportInfo* transport_info =
@@ -137,10 +134,15 @@ bool JsepSessionDescription::AddCandidate(
     updated_candidate.set_password(transport_info->description.ice_pwd);
   }
 
-  candidate_collection_[mediasection_index].add(
-       new JsepIceCandidate(candidate->sdp_mid(),
-                            static_cast<int>(mediasection_index),
-                            updated_candidate));
+  scoped_ptr<JsepIceCandidate> updated_candidate_wrapper(
+      new JsepIceCandidate(candidate->sdp_mid(),
+                           static_cast<int>(mediasection_index),
+                           updated_candidate));
+  if (!candidate_collection_[mediasection_index].HasCandidate(
+          updated_candidate_wrapper.get()))
+    candidate_collection_[mediasection_index].add(
+        updated_candidate_wrapper.release());
+
   return true;
 }
 
diff --git a/talk/app/webrtc/jsepsessiondescription_unittest.cc b/talk/app/webrtc/jsepsessiondescription_unittest.cc
index 671df36d96..55eb3d5392 100644
--- a/talk/app/webrtc/jsepsessiondescription_unittest.cc
+++ b/talk/app/webrtc/jsepsessiondescription_unittest.cc
@@ -204,6 +204,28 @@ TEST_F(JsepSessionDescriptionTest, AddBadCandidate) {
   EXPECT_FALSE(jsep_desc_->AddCandidate(&bad_candidate2));
 }
 
+// Tests that repeatedly adding the same candidate, with or without credentials,
+// does not increase the number of candidates in the description.
+TEST_F(JsepSessionDescriptionTest, AddCandidateDuplicates) {
+  JsepIceCandidate jsep_candidate("", 0, candidate_);
+  EXPECT_TRUE(jsep_desc_->AddCandidate(&jsep_candidate));
+  EXPECT_EQ(1u, jsep_desc_->candidates(0)->count());
+
+  // Add the same candidate again.  It should be ignored.
+  EXPECT_TRUE(jsep_desc_->AddCandidate(&jsep_candidate));
+  EXPECT_EQ(1u, jsep_desc_->candidates(0)->count());
+
+  // Create a new candidate, identical except that the ufrag and pwd are now
+  // populated.
+  candidate_.set_username(kCandidateUfragVoice);
+  candidate_.set_password(kCandidatePwdVoice);
+  JsepIceCandidate jsep_candidate_with_credentials("", 0, candidate_);
+
+  // This should also be identified as redundant and ignored.
+  EXPECT_TRUE(jsep_desc_->AddCandidate(&jsep_candidate_with_credentials));
+  EXPECT_EQ(1u, jsep_desc_->candidates(0)->count());
+}
+
 // Test that we can serialize a JsepSessionDescription and deserialize it again.
 TEST_F(JsepSessionDescriptionTest, SerializeDeserialize) {
   std::string sdp = Serialize(jsep_desc_.get());
diff --git a/talk/app/webrtc/mediastreamhandler.cc b/talk/app/webrtc/mediastreamhandler.cc
index b09af7892f..932d55ee7b 100644
--- a/talk/app/webrtc/mediastreamhandler.cc
+++ b/talk/app/webrtc/mediastreamhandler.cc
@@ -56,14 +56,38 @@ void TrackHandler::OnChanged() {
   }
 }
 
+LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(NULL) {}
+
+LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {}
+
+void LocalAudioSinkAdapter::OnData(const void* audio_data,
+                                   int bits_per_sample,
+                                   int sample_rate,
+                                   int number_of_channels,
+                                   int number_of_frames) {
+  talk_base::CritScope lock(&lock_);
+  if (sink_) {
+    sink_->OnData(audio_data, bits_per_sample, sample_rate,
+                  number_of_channels, number_of_frames);
+  }
+}
+
+void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
+  talk_base::CritScope lock(&lock_);
+  ASSERT(!sink || !sink_);
+  sink_ = sink;
+}
+
 LocalAudioTrackHandler::LocalAudioTrackHandler(
     AudioTrackInterface* track,
     uint32 ssrc,
     AudioProviderInterface* provider)
     : TrackHandler(track, ssrc),
       audio_track_(track),
-      provider_(provider) {
+      provider_(provider),
+      sink_adapter_(new LocalAudioSinkAdapter()) {
   OnEnabledChanged();
+  track->AddSink(sink_adapter_.get());
 }
 
 LocalAudioTrackHandler::~LocalAudioTrackHandler() {
@@ -74,6 +98,7 @@ void LocalAudioTrackHandler::OnStateChanged() {
 }
 
 void LocalAudioTrackHandler::Stop() {
+  audio_track_->RemoveSink(sink_adapter_.get());
   cricket::AudioOptions options;
   provider_->SetAudioSend(ssrc(), false, options, NULL);
 }
@@ -84,8 +109,13 @@ void LocalAudioTrackHandler::OnEnabledChanged() {
     options = static_cast<LocalAudioSource*>(
         audio_track_->GetSource())->options();
   }
-  provider_->SetAudioSend(ssrc(), audio_track_->enabled(), options,
-                          audio_track_->GetRenderer());
+
+  // Use the renderer if the audio track has one, otherwise use the sink
+  // adapter owned by this class.
+  cricket::AudioRenderer* renderer = audio_track_->GetRenderer() ?
+      audio_track_->GetRenderer() : sink_adapter_.get();
+  ASSERT(renderer);
+  provider_->SetAudioSend(ssrc(), audio_track_->enabled(), options, renderer);
 }
 
 RemoteAudioTrackHandler::RemoteAudioTrackHandler(
diff --git a/talk/app/webrtc/mediastreamhandler.h b/talk/app/webrtc/mediastreamhandler.h
index 0cd34d615a..625de85019 100644
--- a/talk/app/webrtc/mediastreamhandler.h
+++ b/talk/app/webrtc/mediastreamhandler.h
@@ -40,6 +40,7 @@
 #include "talk/app/webrtc/mediastreamprovider.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/base/thread.h"
+#include "talk/media/base/audiorenderer.h"
 
 namespace webrtc {
 
@@ -67,6 +68,28 @@ class TrackHandler : public ObserverInterface {
   bool enabled_;
 };
 
+// LocalAudioSinkAdapter receives data callback as a sink to the local
+// AudioTrack, and passes the data to the sink of AudioRenderer.
+class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
+                              public cricket::AudioRenderer {
+ public:
+  LocalAudioSinkAdapter();
+  virtual ~LocalAudioSinkAdapter();
+
+ private:
+  // AudioSinkInterface implementation.
+  virtual void OnData(const void* audio_data, int bits_per_sample,
+                      int sample_rate, int number_of_channels,
+                      int number_of_frames) OVERRIDE;
+
+  // cricket::AudioRenderer implementation.
+  virtual void SetSink(cricket::AudioRenderer::Sink* sink) OVERRIDE;
+
+  cricket::AudioRenderer::Sink* sink_;
+  // Critical section protecting |sink_|.
+  talk_base::CriticalSection lock_;
+};
+
 // LocalAudioTrackHandler listen to events on a local AudioTrack instance
 // connected to a PeerConnection and orders the |provider| to executes the
 // requested change.
@@ -86,6 +109,10 @@ class LocalAudioTrackHandler : public TrackHandler {
  private:
   AudioTrackInterface* audio_track_;
   AudioProviderInterface* provider_;
+
+  // Used to pass the data callback from the |audio_track_| to the other
+  // end of cricket::AudioRenderer.
+  talk_base::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
 };
 
 // RemoteAudioTrackHandler listen to events on a remote AudioTrack instance
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index b2c4468fb9..96d09428ab 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -147,15 +147,33 @@ class VideoTrackInterface : public MediaStreamTrackInterface {
 class AudioSourceInterface : public MediaSourceInterface {
 };
 
+// Interface for receiving audio data from a AudioTrack.
+class AudioTrackSinkInterface {
+ public:
+  virtual void OnData(const void* audio_data,
+                      int bits_per_sample,
+                      int sample_rate,
+                      int number_of_channels,
+                      int number_of_frames) = 0;
+ protected:
+  virtual ~AudioTrackSinkInterface() {}
+};
+
 class AudioTrackInterface : public MediaStreamTrackInterface {
  public:
   // TODO(xians): Figure out if the following interface should be const or not.
   virtual AudioSourceInterface* GetSource() const =  0;
 
+  // Adds/Removes a sink that will receive the audio data from the track.
+  // TODO(xians): Make them pure virtual after Chrome implements these
+  // interfaces.
+  virtual void AddSink(AudioTrackSinkInterface* sink) {}
+  virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
+
   // Gets a pointer to the audio renderer of this AudioTrack.
   // The pointer is valid for the lifetime of this AudioTrack.
-  // TODO(xians): Make the following interface pure virtual once Chrome has its
-  // implementation.
+  // TODO(xians): Remove the following interface after Chrome switches to
+  // AddSink() and RemoveSink() interfaces.
   virtual cricket::AudioRenderer* GetRenderer() { return NULL; }
 
  protected:
diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index f9d5e2eee8..98e0645839 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -180,7 +180,6 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
     EXPECT_EQ(pc_->ice_gathering_state(), new_state);
   }
   virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
-
     EXPECT_NE(PeerConnectionInterface::kIceGatheringNew,
               pc_->ice_gathering_state());
 
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 0de46e71bf..59d72709f3 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -786,7 +786,7 @@ bool WebRtcSession::ProcessIceMessage(const IceCandidateInterface* candidate) {
     return false;
   }
 
-  return UseCandidatesInSessionDescription(remote_desc_.get());
+  return UseCandidate(candidate);
 }
 
 bool WebRtcSession::GetTrackIdBySsrc(uint32 ssrc, std::string* id) {
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index 6d3a58131e..ba58936e39 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -1421,10 +1421,12 @@ TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
   EXPECT_EQ(1, candidates->at(0)->candidate().component());
   EXPECT_EQ(2, candidates->at(1)->candidate().component());
 
+  // |ice_candidate3| is identical to |ice_candidate2|.  It can be added
+  // successfully, but the total count of candidates will not increase.
   candidate.set_component(2);
   JsepIceCandidate ice_candidate3(kMediaContentName0, 0, candidate);
   EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate3));
-  ASSERT_EQ(3u, candidates->count());
+  ASSERT_EQ(2u, candidates->count());
 
   JsepIceCandidate bad_ice_candidate("bad content name", 99, candidate);
   EXPECT_FALSE(session_->ProcessIceMessage(&bad_ice_candidate));