Socket options are only applied when first setting TransportChannelImpl.

Also fixed the issue when we have an TransportChannelImpl, the socket
option is not preserved.

Since this is a code path that will be modified by bundle (which Peter also has a test case already), we don't need a test case here.

BUG=4374
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42699004

Cr-Commit-Position: refs/heads/master@{#8702}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8702 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/app/webrtc/webrtcsession_unittest.cc

@@ -3546,6 +3546,54 @@ TEST_F(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) {
   SetLocalDescriptionWithoutError(answer);
 }
 
+// This verifies that the voice channel after bundle has both options from video
+// and voice channels.
+TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) {
+  InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
+  mediastream_signaling_.SendAudioVideoStream1();
+
+  PeerConnectionInterface::RTCOfferAnswerOptions options;
+  options.use_rtp_mux = true;
+
+  SessionDescriptionInterface* offer = CreateOffer(options);
+  SetLocalDescriptionWithoutError(offer);
+
+  session_->video_channel()->SetOption(cricket::BaseChannel::ST_RTP,
+                                       rtc::Socket::Option::OPT_SNDBUF, 4000);
+
+  session_->voice_channel()->SetOption(cricket::BaseChannel::ST_RTP,
+                                       rtc::Socket::Option::OPT_RCVBUF, 8000);
+
+  int option_val;
+  EXPECT_TRUE(session_->video_channel()->transport_channel()->GetOption(
+      rtc::Socket::Option::OPT_SNDBUF, &option_val));
+  EXPECT_EQ(4000, option_val);
+  EXPECT_FALSE(session_->voice_channel()->transport_channel()->GetOption(
+      rtc::Socket::Option::OPT_SNDBUF, &option_val));
+
+  EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
+      rtc::Socket::Option::OPT_RCVBUF, &option_val));
+  EXPECT_EQ(8000, option_val);
+  EXPECT_FALSE(session_->video_channel()->transport_channel()->GetOption(
+      rtc::Socket::Option::OPT_RCVBUF, &option_val));
+
+  EXPECT_NE(session_->voice_channel()->transport_channel(),
+            session_->video_channel()->transport_channel());
+
+  mediastream_signaling_.SendAudioVideoStream2();
+  SessionDescriptionInterface* answer =
+      CreateRemoteAnswer(session_->local_description());
+  SetRemoteDescriptionWithoutError(answer);
+
+  EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
+      rtc::Socket::Option::OPT_SNDBUF, &option_val));
+  EXPECT_EQ(4000, option_val);
+
+  EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
+      rtc::Socket::Option::OPT_RCVBUF, &option_val));
+  EXPECT_EQ(8000, option_val);
+}
+
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.

talk/session/media/channel.h

@@ -230,6 +230,9 @@ class BaseChannel
   // Made public for easier testing.
   void SetReadyToSend(TransportChannel* channel, bool ready);
 
+  // Only public for unit tests.  Otherwise, consider protected.
+  virtual int SetOption(SocketType type, rtc::Socket::Option o, int val);
+
  protected:
   MediaEngineInterface* media_engine() const { return media_engine_; }
   virtual MediaChannel* media_channel() const { return media_channel_; }
@@ -254,7 +257,6 @@ class BaseChannel
                           rtc::DiffServCodePoint dscp);
   virtual bool SendRtcp(rtc::Buffer* packet,
                         rtc::DiffServCodePoint dscp);
-  virtual int SetOption(SocketType type, rtc::Socket::Option o, int val);
 
   // From TransportChannel
   void OnWritableState(TransportChannel* channel);

webrtc/p2p/base/transportchannelproxy.cc

@@ -67,17 +67,14 @@ void TransportChannelProxy::SetImplementation(TransportChannelImpl* impl) {
         this, &TransportChannelProxy::OnReadyToSend);
     impl_->SignalRouteChange.connect(
         this, &TransportChannelProxy::OnRouteChange);
-    for (OptionList::iterator it = pending_options_.begin();
-         it != pending_options_.end();
-         ++it) {
-      impl_->SetOption(it->first, it->second);
+    for (const auto& pair : options_) {
+      impl_->SetOption(pair.first, pair.second);
     }
 
     // Push down the SRTP ciphers, if any were set.
     if (!pending_srtp_ciphers_.empty()) {
       impl_->SetSrtpCiphers(pending_srtp_ciphers_);
     }
-    pending_options_.clear();
   }
 
   // Post ourselves a message to see if we need to fire state callbacks.
@@ -97,8 +94,8 @@ int TransportChannelProxy::SendPacket(const char* data, size_t len,
 
 int TransportChannelProxy::SetOption(rtc::Socket::Option opt, int value) {
   ASSERT(rtc::Thread::Current() == worker_thread_);
+  options_.push_back(OptionPair(opt, value));
   if (!impl_) {
-    pending_options_.push_back(OptionPair(opt, value));
     return 0;
   }
   return impl_->SetOption(opt, value);
@@ -110,9 +107,9 @@ bool TransportChannelProxy::GetOption(rtc::Socket::Option opt, int* value) {
     return impl_->GetOption(opt, value);
   }
 
-  for (const auto& pending : pending_options_) {
-    if (pending.first == opt) {
-      *value = pending.second;
+  for (const auto& pair : options_) {
+    if (pair.first == opt) {
+      *value = pair.second;
       return true;
     }
   }

webrtc/p2p/base/transportchannelproxy.h

@@ -88,7 +88,7 @@ class TransportChannelProxy : public TransportChannel,
   std::string name_;
   rtc::Thread* worker_thread_;
   TransportChannelImpl* impl_;
-  OptionList pending_options_;
+  OptionList options_;
   std::vector<std::string> pending_srtp_ciphers_;
 
   DISALLOW_EVIL_CONSTRUCTORS(TransportChannelProxy);