Update libjingle to CL 53496343.
Review URL: https://webrtc-codereview.appspot.com/2323005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4882 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/mediaconstraintsinterface.h b/talk/app/webrtc/mediaconstraintsinterface.h
index bc3872c..48022dd 100644
--- a/talk/app/webrtc/mediaconstraintsinterface.h
+++ b/talk/app/webrtc/mediaconstraintsinterface.h
@@ -112,10 +112,13 @@
// stripped by Chrome before passed down to Libjingle.
static const char kInternalConstraintPrefix[];
- // This constraint is for internal use only, representing the Chrome command
- // line flag. So it is prefixed with "internal" so JS values will be removed.
+ // These constraints are for internal use only, representing Chrome command
+ // line flags. So they are prefixed with "internal" so JS values will be
+ // removed.
// Used by a local audio source.
static const char kInternalAecDump[]; // internalAecDump
+ // Used for disabling security and use plain RTP.
+ static const char kInternalDisableEncryption[]; // internalDisableEncryption
protected:
// Dtor protected as objects shouldn't be deleted via this interface
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 83f8912..ff331a9 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -69,6 +69,9 @@
const char MediaConstraintsInterface::kEnableSctpDataChannels[] =
"internalSctpDataChannels";
+const char MediaConstraintsInterface::kInternalDisableEncryption[] =
+ "internalDisableEncryption";
+
// Error messages
const char kSetLocalSdpFailed[] = "SetLocalDescription failed: ";
const char kSetRemoteSdpFailed[] = "SetRemoteDescription failed: ";
@@ -488,6 +491,15 @@
webrtc_session_desc_factory_->SignalIdentityReady.connect(
this, &WebRtcSession::OnIdentityReady);
+
+ // Disable encryption if kDisableEncryption is set.
+ if (FindConstraint(
+ constraints,
+ MediaConstraintsInterface::kInternalDisableEncryption,
+ &value, NULL) && value) {
+ webrtc_session_desc_factory_->set_secure(cricket::SEC_DISABLED);
+ }
+
return true;
}
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index acbc924..c47fb17 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -1877,8 +1877,10 @@
}
TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
+ constraints_.reset(new FakeConstraints());
+ constraints_->AddOptional(
+ webrtc::MediaConstraintsInterface::kInternalDisableEncryption, true);
Init(NULL);
- session_->set_secure_policy(cricket::SEC_DISABLED);
mediastream_signaling_.SendAudioVideoStream1();
scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
@@ -2342,9 +2344,11 @@
// This test verifies the crypto parameter when security is disabled.
TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
+ constraints_.reset(new FakeConstraints());
+ constraints_->AddOptional(
+ webrtc::MediaConstraintsInterface::kInternalDisableEncryption, true);
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
- session_->set_secure_policy(cricket::SEC_DISABLED);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
diff --git a/talk/base/natsocketfactory.cc b/talk/base/natsocketfactory.cc
index a7c4240..1686f20 100644
--- a/talk/base/natsocketfactory.cc
+++ b/talk/base/natsocketfactory.cc
@@ -85,7 +85,7 @@
class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
public:
explicit NATSocket(NATInternalSocketFactory* sf, int family, int type)
- : sf_(sf), family_(family), type_(type), async_(true), connected_(false),
+ : sf_(sf), family_(family), type_(type), connected_(false),
socket_(NULL), buf_(NULL), size_(0) {
}
@@ -312,7 +312,6 @@
NATInternalSocketFactory* sf_;
int family_;
int type_;
- bool async_;
bool connected_;
SocketAddress remote_addr_;
SocketAddress server_addr_; // address of the NAT server
diff --git a/talk/media/base/rtpdataengine_unittest.cc b/talk/media/base/rtpdataengine_unittest.cc
index 37ea968..bc46818 100644
--- a/talk/media/base/rtpdataengine_unittest.cc
+++ b/talk/media/base/rtpdataengine_unittest.cc
@@ -299,8 +299,7 @@
EXPECT_EQ(header0.timestamp + 180000, header1.timestamp);
}
-// See: https://code.google.com/p/webrtc/issues/detail?id=2409
-TEST_F(RtpDataMediaChannelTest, DISABLED_SendDataMultipleClocks) {
+TEST_F(RtpDataMediaChannelTest, SendDataMultipleClocks) {
// Timings owned by RtpDataEngines.
FakeTiming* timing1 = new FakeTiming();
talk_base::scoped_ptr<cricket::RtpDataEngine> dme1(CreateEngine(timing1));
diff --git a/talk/media/webrtc/webrtcmediaengine.h b/talk/media/webrtc/webrtcmediaengine.h
index a2ee658..5690ca0 100644
--- a/talk/media/webrtc/webrtcmediaengine.h
+++ b/talk/media/webrtc/webrtcmediaengine.h
@@ -87,10 +87,13 @@
virtual SoundclipMedia* CreateSoundclip() OVERRIDE {
return delegate_->CreateSoundclip();
}
- virtual bool SetAudioOptions(int options) OVERRIDE {
+ virtual AudioOptions GetAudioOptions() const OVERRIDE {
+ return delegate_->GetAudioOptions();
+ }
+ virtual bool SetAudioOptions(const AudioOptions& options) OVERRIDE {
return delegate_->SetAudioOptions(options);
}
- virtual bool SetVideoOptions(int options) OVERRIDE {
+ virtual bool SetVideoOptions(const VideoOptions& options) OVERRIDE {
return delegate_->SetVideoOptions(options);
}
virtual bool SetAudioDelayOffset(int offset) OVERRIDE {
