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
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index b23ba54..6bfa17e4 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -3546,6 +3546,54 @@
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.
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index 05818a0..1d0a887 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -230,6 +230,9 @@
// 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 @@
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);
diff --git a/webrtc/p2p/base/transportchannelproxy.cc b/webrtc/p2p/base/transportchannelproxy.cc
index a278d96..5118920 100644
--- a/webrtc/p2p/base/transportchannelproxy.cc
+++ b/webrtc/p2p/base/transportchannelproxy.cc
@@ -67,17 +67,14 @@
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::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 @@
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;
}
}
diff --git a/webrtc/p2p/base/transportchannelproxy.h b/webrtc/p2p/base/transportchannelproxy.h
index 19870028..ffdad43 100644
--- a/webrtc/p2p/base/transportchannelproxy.h
+++ b/webrtc/p2p/base/transportchannelproxy.h
@@ -88,7 +88,7 @@
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);
