Add DTLSTransport info into sender/receiver state.
This is in preparation for letting Chrome extract DTLSTransport
information after SLD/SRD instead of doing it on-demand.
Bug: chromium:907849
Change-Id: Iac6b174c98d3d14136e1fd25bce4a9292f6c8b41
Reviewed-on: https://webrtc-review.googlesource.com/c/116984
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26289}diff --git a/api/dtls_transport_interface.h b/api/dtls_transport_interface.h
index abe7378..1faf3f5 100644
--- a/api/dtls_transport_interface.h
+++ b/api/dtls_transport_interface.h
@@ -23,7 +23,9 @@
kConnecting, // In the process of negotiating a secure connection.
kConnected, // Completed negotiation and verified fingerprints.
kClosed, // Intentionally closed.
- kFailed // Failure due to an error or failing to verify a remote fingerprint.
+ kFailed, // Failure due to an error or failing to verify a remote
+ // fingerprint.
+ kNumValues
};
// This object gives snapshot information about the changeable state of a
diff --git a/api/rtp_receiver_interface.cc b/api/rtp_receiver_interface.cc
index d8bb3d3..52f72df 100644
--- a/api/rtp_receiver_interface.cc
+++ b/api/rtp_receiver_interface.cc
@@ -53,4 +53,9 @@
return nullptr;
}
+rtc::scoped_refptr<DtlsTransportInterface>
+RtpReceiverInterface::dtls_transport() const {
+ return nullptr;
+}
+
} // namespace webrtc
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index 12c9d95..e7fa0bf 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -18,6 +18,7 @@
#include <vector>
#include "api/crypto/frame_decryptor_interface.h"
+#include "api/dtls_transport_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/proxy.h"
@@ -92,6 +93,13 @@
class RtpReceiverInterface : public rtc::RefCountInterface {
public:
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+
+ // The dtlsTransport attribute exposes the DTLS transport on which the
+ // media is received. It may be null.
+ // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-transport
+ // TODO(https://bugs.webrtc.org/907849) remove default implementation
+ virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
+
// The list of streams that |track| is associated with. This is the same as
// the [[AssociatedRemoteMediaStreams]] internal slot in the spec.
// https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams
@@ -146,6 +154,7 @@
BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
PROXY_SIGNALING_THREAD_DESTRUCTOR()
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
streams)
diff --git a/api/rtp_sender_interface.cc b/api/rtp_sender_interface.cc
index 68747ea..d23fd18 100644
--- a/api/rtp_sender_interface.cc
+++ b/api/rtp_sender_interface.cc
@@ -25,4 +25,9 @@
return {};
}
+rtc::scoped_refptr<DtlsTransportInterface> RtpSenderInterface::dtls_transport()
+ const {
+ return nullptr;
+}
+
} // namespace webrtc
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index c1dc716..6397938 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -18,6 +18,7 @@
#include <vector>
#include "api/crypto/frame_encryptor_interface.h"
+#include "api/dtls_transport_interface.h"
#include "api/dtmf_sender_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
@@ -36,6 +37,12 @@
virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+ // The dtlsTransport attribute exposes the DTLS transport on which the
+ // media is sent. It may be null.
+ // https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport
+ // TODO(https://bugs.webrtc.org/907849) remove default implementation
+ virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
+
// Returns primary SSRC used by this sender for sending media.
// Returns 0 if not yet determined.
// TODO(deadbeef): Change to absl::optional.
@@ -91,6 +98,7 @@
PROXY_SIGNALING_THREAD_DESTRUCTOR()
PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
PROXY_CONSTMETHOD0(uint32_t, ssrc)
PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index bc7b3c0..c39833b 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -421,6 +421,7 @@
deps = [
":rtc_event_log_api",
"../api:libjingle_logging_api",
+ "../api:libjingle_peerconnection_api",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/memory",
]
diff --git a/logging/rtc_event_log/events/rtc_event_dtls_transport_state.h b/logging/rtc_event_log/events/rtc_event_dtls_transport_state.h
index e76cfe9..b61cb33 100644
--- a/logging/rtc_event_log/events/rtc_event_dtls_transport_state.h
+++ b/logging/rtc_event_log/events/rtc_event_dtls_transport_state.h
@@ -13,19 +13,11 @@
#include <memory>
+#include "api/dtls_transport_interface.h"
#include "logging/rtc_event_log/events/rtc_event.h"
namespace webrtc {
-enum class DtlsTransportState {
- kNew,
- kConnecting,
- kConnected,
- kClosed,
- kFailed,
- kNumValues
-};
-
class RtcEventDtlsTransportState : public RtcEvent {
public:
explicit RtcEventDtlsTransportState(DtlsTransportState state);
diff --git a/pc/jsep_transport.h b/pc/jsep_transport.h
index 648fe29..21747a6 100644
--- a/pc/jsep_transport.h
+++ b/pc/jsep_transport.h
@@ -189,7 +189,7 @@
}
}
- rtc::scoped_refptr<webrtc::DtlsTransportInterface> RtpDtlsTransport() {
+ rtc::scoped_refptr<webrtc::DtlsTransport> RtpDtlsTransport() {
return rtp_dtls_transport_;
}
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 926d71d..a7d1fd7 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -173,7 +173,7 @@
return jsep_transport->rtcp_dtls_transport();
}
-rtc::scoped_refptr<webrtc::DtlsTransportInterface>
+rtc::scoped_refptr<webrtc::DtlsTransport>
JsepTransportController::LookupDtlsTransportByMid(const std::string& mid) {
auto jsep_transport = GetJsepTransportForMid(mid);
if (!jsep_transport) {
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index e57b7d8..e160be3 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@@ -117,7 +117,7 @@
const cricket::DtlsTransportInternal* GetRtcpDtlsTransport(
const std::string& mid) const;
// Gets the externally sharable version of the DtlsTransport.
- rtc::scoped_refptr<webrtc::DtlsTransportInterface> LookupDtlsTransportByMid(
+ rtc::scoped_refptr<webrtc::DtlsTransport> LookupDtlsTransportByMid(
const std::string& mid);
MediaTransportInterface* GetMediaTransport(const std::string& mid) const;
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 2d68c81..41fadd8 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -2106,6 +2106,18 @@
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
for (auto transceiver : transceivers_) {
+ // 2.2.7.1.1.(6-9): Set sender and receiver's transport slots.
+ // Note that code paths that don't set MID won't be able to use
+ // information about DTLS transports.
+ if (transceiver->mid()) {
+ auto dtls_transport =
+ LookupDtlsTransportByMidInternal(*transceiver->mid());
+ transceiver->internal()->sender_internal()->set_transport(
+ dtls_transport);
+ transceiver->internal()->receiver_internal()->set_transport(
+ dtls_transport);
+ }
+
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
@@ -2548,7 +2560,7 @@
now_receiving_transceivers.push_back(transceiver);
}
}
- // 2.2.8.1.7: If direction is "sendonly" or "inactive", and transceiver's
+ // 2.2.8.1.9: If direction is "sendonly" or "inactive", and transceiver's
// [[FiredDirection]] slot is either "sendrecv" or "recvonly", process the
// removal of a remote track for the media description, given transceiver,
// removeList, and muteTracks.
@@ -2558,16 +2570,25 @@
ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
&removed_streams);
}
- // 2.2.8.1.8: Set transceiver's [[FiredDirection]] slot to direction.
+ // 2.2.8.1.10: Set transceiver's [[FiredDirection]] slot to direction.
transceiver->internal()->set_fired_direction(local_direction);
- // 2.2.8.1.9: If description is of type "answer" or "pranswer", then run
+ // 2.2.8.1.11: If description is of type "answer" or "pranswer", then run
// the following steps:
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
- // 2.2.8.1.9.1: Set transceiver's [[CurrentDirection]] slot to
+ // 2.2.8.1.11.1: Set transceiver's [[CurrentDirection]] slot to
// direction.
transceiver->internal()->set_current_direction(local_direction);
+ // 2.2.8.1.11.[3-6]: Set the transport internal slots.
+ if (transceiver->mid()) {
+ auto dtls_transport =
+ LookupDtlsTransportByMidInternal(*transceiver->mid());
+ transceiver->internal()->sender_internal()->set_transport(
+ dtls_transport);
+ transceiver->internal()->receiver_internal()->set_transport(
+ dtls_transport);
+ }
}
- // 2.2.8.1.10: If the media description is rejected, and transceiver is
+ // 2.2.8.1.12: If the media description is rejected, and transceiver is
// not already stopped, stop the RTCRtpTransceiver transceiver.
if (content->rejected && !transceiver->stopped()) {
RTC_LOG(LS_INFO) << "Stopping transceiver for MID=" << content->name
@@ -3445,6 +3466,11 @@
return transport_controller_->LookupDtlsTransportByMid(mid);
}
+rtc::scoped_refptr<DtlsTransport>
+PeerConnection::LookupDtlsTransportByMidInternal(const std::string& mid) {
+ return transport_controller_->LookupDtlsTransportByMid(mid);
+}
+
const SessionDescriptionInterface* PeerConnection::local_description() const {
return pending_local_description_ ? pending_local_description_.get()
: current_local_description_.get();
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index b11c867..ca92d76 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -197,6 +197,8 @@
rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
const std::string& mid) override;
+ rtc::scoped_refptr<DtlsTransport> LookupDtlsTransportByMidInternal(
+ const std::string& mid);
RTC_DEPRECATED bool StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) override;
diff --git a/pc/peer_connection_signaling_unittest.cc b/pc/peer_connection_signaling_unittest.cc
index ed76753..340d472 100644
--- a/pc/peer_connection_signaling_unittest.cc
+++ b/pc/peer_connection_signaling_unittest.cc
@@ -108,6 +108,28 @@
return wrapper;
}
+ int NumberOfDtlsTransports(const WrapperPtr& pc_wrapper) {
+ std::set<DtlsTransportInterface*> transports;
+ auto transceivers = pc_wrapper->pc()->GetTransceivers();
+
+ for (auto& transceiver : transceivers) {
+ if (transceiver->sender()->dtls_transport()) {
+ EXPECT_TRUE(transceiver->receiver()->dtls_transport());
+ EXPECT_EQ(transceiver->sender()->dtls_transport().get(),
+ transceiver->receiver()->dtls_transport().get());
+ transports.insert(transceiver->sender()->dtls_transport().get());
+ } else {
+ // If one transceiver is missing, they all should be.
+ EXPECT_EQ(0UL, transports.size());
+ }
+ }
+ return transports.size();
+ }
+
+ bool HasDtlsTransport(const WrapperPtr& pc_wrapper) {
+ return NumberOfDtlsTransports(pc_wrapper) > 0;
+ }
+
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
@@ -505,4 +527,74 @@
Values(SdpSemantics::kPlanB,
SdpSemantics::kUnifiedPlan));
+class PeerConnectionSignalingUnifiedPlanTest
+ : public PeerConnectionSignalingBaseTest {
+ protected:
+ PeerConnectionSignalingUnifiedPlanTest()
+ : PeerConnectionSignalingBaseTest(SdpSemantics::kUnifiedPlan) {}
+};
+
+// Test that transports are shown in the sender/receiver API after offer/answer.
+// This only works in Unified Plan.
+TEST_F(PeerConnectionSignalingUnifiedPlanTest,
+ DtlsTransportsInstantiateInOfferAnswer) {
+ auto caller = CreatePeerConnectionWithAudioVideo();
+ auto callee = CreatePeerConnection();
+
+ EXPECT_FALSE(HasDtlsTransport(caller));
+ EXPECT_FALSE(HasDtlsTransport(callee));
+ auto offer = caller->CreateOffer(RTCOfferAnswerOptions());
+ caller->SetLocalDescription(CloneSessionDescription(offer.get()));
+ EXPECT_TRUE(HasDtlsTransport(caller));
+ callee->SetRemoteDescription(std::move(offer));
+ EXPECT_FALSE(HasDtlsTransport(callee));
+ auto answer = callee->CreateAnswer(RTCOfferAnswerOptions());
+ callee->SetLocalDescription(CloneSessionDescription(answer.get()));
+ EXPECT_TRUE(HasDtlsTransport(callee));
+ caller->SetRemoteDescription(std::move(answer));
+ EXPECT_TRUE(HasDtlsTransport(caller));
+
+ ASSERT_EQ(SignalingState::kStable, caller->signaling_state());
+}
+
+TEST_F(PeerConnectionSignalingUnifiedPlanTest, DtlsTransportsMergeWhenBundled) {
+ auto caller = CreatePeerConnectionWithAudioVideo();
+ auto callee = CreatePeerConnection();
+
+ EXPECT_FALSE(HasDtlsTransport(caller));
+ EXPECT_FALSE(HasDtlsTransport(callee));
+ auto offer = caller->CreateOffer(RTCOfferAnswerOptions());
+ caller->SetLocalDescription(CloneSessionDescription(offer.get()));
+ EXPECT_EQ(2, NumberOfDtlsTransports(caller));
+ callee->SetRemoteDescription(std::move(offer));
+ auto answer = callee->CreateAnswer(RTCOfferAnswerOptions());
+ callee->SetLocalDescription(CloneSessionDescription(answer.get()));
+ caller->SetRemoteDescription(std::move(answer));
+ EXPECT_EQ(1, NumberOfDtlsTransports(caller));
+
+ ASSERT_EQ(SignalingState::kStable, caller->signaling_state());
+}
+
+TEST_F(PeerConnectionSignalingUnifiedPlanTest,
+ DtlsTransportsAreSeparateeWhenUnbundled) {
+ auto caller = CreatePeerConnectionWithAudioVideo();
+ auto callee = CreatePeerConnection();
+
+ EXPECT_FALSE(HasDtlsTransport(caller));
+ EXPECT_FALSE(HasDtlsTransport(callee));
+ RTCOfferAnswerOptions unbundle_options;
+ unbundle_options.use_rtp_mux = false;
+ auto offer = caller->CreateOffer(unbundle_options);
+ caller->SetLocalDescription(CloneSessionDescription(offer.get()));
+ EXPECT_EQ(2, NumberOfDtlsTransports(caller));
+ callee->SetRemoteDescription(std::move(offer));
+ auto answer = callee->CreateAnswer(RTCOfferAnswerOptions());
+ callee->SetLocalDescription(CloneSessionDescription(answer.get()));
+ EXPECT_EQ(2, NumberOfDtlsTransports(callee));
+ caller->SetRemoteDescription(std::move(answer));
+ EXPECT_EQ(2, NumberOfDtlsTransports(caller));
+
+ ASSERT_EQ(SignalingState::kStable, caller->signaling_state());
+}
+
} // namespace webrtc
diff --git a/pc/rtp_receiver.h b/pc/rtp_receiver.h
index 754fce0..86673e8 100644
--- a/pc/rtp_receiver.h
+++ b/pc/rtp_receiver.h
@@ -54,6 +54,8 @@
// receive packets on unsignaled SSRCs.
virtual void SetupMediaChannel(uint32_t ssrc) = 0;
+ virtual void set_transport(
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
// This SSRC is used as an identifier for the receiver between the API layer
// and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
virtual uint32_t ssrc() const = 0;
@@ -106,6 +108,9 @@
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
+ return dtls_transport_;
+ }
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
@@ -133,6 +138,10 @@
uint32_t ssrc() const override { return ssrc_.value_or(0); }
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
+ void set_transport(
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
+ dtls_transport_ = dtls_transport;
+ }
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
@@ -160,6 +169,7 @@
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
};
class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
@@ -186,6 +196,9 @@
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
+ return dtls_transport_;
+ }
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
@@ -213,6 +226,10 @@
uint32_t ssrc() const override { return ssrc_.value_or(0); }
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
+ void set_transport(
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
+ dtls_transport_ = dtls_transport;
+ }
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
@@ -257,6 +274,7 @@
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
};
} // namespace webrtc
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index 6ed0603..fd8980c 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -50,6 +50,8 @@
virtual void set_stream_ids(const std::vector<std::string>& stream_ids) = 0;
virtual void set_init_send_encodings(
const std::vector<RtpEncodingParameters>& init_send_encodings) = 0;
+ virtual void set_transport(
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
virtual void Stop() = 0;
@@ -112,6 +114,10 @@
return track_;
}
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
+ return dtls_transport_;
+ }
+
uint32_t ssrc() const override { return ssrc_; }
cricket::MediaType media_type() const override {
@@ -147,7 +153,10 @@
std::vector<RtpEncodingParameters> init_send_encodings() const override {
return init_parameters_.encodings;
}
-
+ void set_transport(
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
+ dtls_transport_ = dtls_transport;
+ }
void Stop() override;
int AttachmentId() const override { return attachment_id_; }
@@ -173,6 +182,7 @@
cricket::VoiceMediaChannel* media_channel_ = nullptr;
StatsCollector* stats_ = nullptr;
rtc::scoped_refptr<AudioTrackInterface> track_;
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
absl::optional<std::string> last_transaction_id_;
uint32_t ssrc_ = 0;
@@ -205,6 +215,9 @@
}
uint32_t ssrc() const override { return ssrc_; }
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
+ return dtls_transport_;
+ }
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
@@ -221,6 +234,10 @@
std::vector<RtpEncodingParameters> init_send_encodings() const override {
return init_parameters_.encodings;
}
+ void set_transport(
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
+ dtls_transport_ = dtls_transport;
+ }
RtpParameters GetParameters() override;
RTCError SetParameters(const RtpParameters& parameters) override;
@@ -266,6 +283,7 @@
bool stopped_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_;
+ rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
};
} // namespace webrtc
diff --git a/pc/test/mock_rtp_receiver_internal.h b/pc/test/mock_rtp_receiver_internal.h
index 10807da..f948d83 100644
--- a/pc/test/mock_rtp_receiver_internal.h
+++ b/pc/test/mock_rtp_receiver_internal.h
@@ -25,6 +25,8 @@
// RtpReceiverInterface methods.
MOCK_METHOD1(SetTrack, void(MediaStreamTrackInterface*));
MOCK_CONST_METHOD0(track, rtc::scoped_refptr<MediaStreamTrackInterface>());
+ MOCK_CONST_METHOD0(dtls_transport,
+ rtc::scoped_refptr<DtlsTransportInterface>());
MOCK_CONST_METHOD0(stream_ids, std::vector<std::string>());
MOCK_CONST_METHOD0(streams,
std::vector<rtc::scoped_refptr<MediaStreamInterface>>());
@@ -46,6 +48,7 @@
MOCK_CONST_METHOD0(ssrc, uint32_t());
MOCK_METHOD0(NotifyFirstPacketReceived, void());
MOCK_METHOD1(set_stream_ids, void(std::vector<std::string>));
+ MOCK_METHOD1(set_transport, void(rtc::scoped_refptr<DtlsTransportInterface>));
MOCK_METHOD1(
SetStreams,
void(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&));
diff --git a/pc/test/mock_rtp_sender_internal.h b/pc/test/mock_rtp_sender_internal.h
index 7859175..fff81f4 100644
--- a/pc/test/mock_rtp_sender_internal.h
+++ b/pc/test/mock_rtp_sender_internal.h
@@ -26,10 +26,13 @@
MOCK_METHOD1(SetTrack, bool(MediaStreamTrackInterface*));
MOCK_CONST_METHOD0(track, rtc::scoped_refptr<MediaStreamTrackInterface>());
MOCK_CONST_METHOD0(ssrc, uint32_t());
+ MOCK_CONST_METHOD0(dtls_transport,
+ rtc::scoped_refptr<DtlsTransportInterface>());
MOCK_CONST_METHOD0(media_type, cricket::MediaType());
MOCK_CONST_METHOD0(id, std::string());
MOCK_CONST_METHOD0(stream_ids, std::vector<std::string>());
MOCK_CONST_METHOD0(init_send_encodings, std::vector<RtpEncodingParameters>());
+ MOCK_METHOD1(set_transport, void(rtc::scoped_refptr<DtlsTransportInterface>));
MOCK_METHOD0(GetParameters, RtpParameters());
MOCK_METHOD1(SetParameters, RTCError(const RtpParameters&));
MOCK_CONST_METHOD0(GetDtmfSender, rtc::scoped_refptr<DtmfSenderInterface>());