Revert "Create RTCDtlsTransport objects in the blink layer"

This reverts commit d7d59c5b.

Reason for revert: virtual/webrtc-wpt-unified-plan/external/wpt/webrtc/RTCRtpSender-transport.https.html fails in "WebKit Linux Trusty MSAN".

Original change's description:
> Create RTCDtlsTransport objects in the blink layer
> 
> This CL does on-demand creation of blink-layer transport
> objects. It also introduces a way to get at the webrtc::PeerConnection
> object from the WebPeerConnectionHandler API.
> 
> Bug: chromium:907849
> Change-Id: Ic86a5a66289c31d1f939a47d58bc194c32471cbd
> Reviewed-on: https://chromium-review.googlesource.com/c/1358060
> Commit-Queue: Harald Alvestrand <hta@chromium.org>
> Reviewed-by: Guido Urdaneta <guidou@chromium.org>
> Reviewed-by: Henrik Boström <hbos@chromium.org>
> Reviewed-by: Kentaro Hara <haraken@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#615072}

TBR=hta@chromium.org,haraken@chromium.org,hbos@chromium.org,guidou@chromium.org

Change-Id: Ie9c8d2e3dc5f97fc94ca5bf43cc30a9407d9fa4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:907849
Reviewed-on: https://chromium-review.googlesource.com/c/1369719Reviewed-by: Samuel Huang <huangs@chromium.org>
Commit-Queue: Samuel Huang <huangs@chromium.org>
Cr-Commit-Position: refs/heads/master@{#615115}

content/renderer/media/webrtc/mock_peer_connection_impl.h

@@ -13,7 +13,6 @@
 #include "base/macros.h"
 #include "base/optional.h"
 #include "testing/gmock/include/gmock/gmock.h"
-#include "third_party/webrtc/api/dtlstransportinterface.h"
 #include "third_party/webrtc/api/peerconnectioninterface.h"
 #include "third_party/webrtc/api/stats/rtcstatsreport.h"
 
@@ -158,10 +157,6 @@ class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface {
   void SetGetStatsResult(bool result) { getstats_result_ = result; }
   // Set the report that |GetStats(RTCStatsCollectorCallback*)| returns.
   void SetGetStatsReport(webrtc::RTCStatsReport* report);
-  rtc::scoped_refptr<webrtc::DtlsTransportInterface> LookupDtlsTransportByMid(
-      const std::string& mid) override {
-    return nullptr;
-  }
 
   SignalingState signaling_state() override {
     NOTIMPLEMENTED();

content/renderer/media/webrtc/rtc_peer_connection_handler.cc

@@ -1924,11 +1924,6 @@ blink::WebString RTCPeerConnectionHandler::Id() const {
   return blink::WebString::FromASCII(id_);
 }
 
-webrtc::PeerConnectionInterface*
-RTCPeerConnectionHandler::NativePeerConnection() {
-  return native_peer_connection();
-}
-
 void RTCPeerConnectionHandler::OnSignalingChange(
     webrtc::PeerConnectionInterface::SignalingState new_state) {
   DCHECK(task_runner_->RunsTasksInCurrentSequence());

content/renderer/media/webrtc/rtc_peer_connection_handler.h

@@ -170,7 +170,6 @@ class CONTENT_EXPORT RTCPeerConnectionHandler
       const blink::WebRTCDataChannelInit& init) override;
   void Stop() override;
   blink::WebString Id() const override;
-  webrtc::PeerConnectionInterface* NativePeerConnection() override;
 
   // Delegate functions to allow for mocking of WebKit interfaces.
   // getStats takes ownership of request parameter.

third_party/blink/public/platform/web_rtc_peer_connection_handler.h

@@ -131,9 +131,6 @@ class WebRTCPeerConnectionHandler {
 
   // Origin Trial - RtcPeerConnectionId
   virtual WebString Id() const = 0;
-
-  // Returns a pointer to the underlying native PeerConnection object.
-  virtual webrtc::PeerConnectionInterface* NativePeerConnection() = 0;
 };
 
 }  // namespace blink

third_party/blink/renderer/modules/peerconnection/rtc_dtls_transport.cc

@@ -21,7 +21,6 @@
 #include "third_party/blink/renderer/modules/peerconnection/rtc_peer_connection_ice_event_init.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_quic_transport.h"
 #include "third_party/blink/renderer/platform/scheduler/public/thread.h"
-#include "third_party/webrtc/api/dtlstransportinterface.h"
 #include "third_party/webrtc/api/jsepicecandidate.h"
 #include "third_party/webrtc/api/peerconnectioninterface.h"
 #include "third_party/webrtc/p2p/base/portallocator.h"
@@ -31,10 +30,12 @@
 
 namespace blink {
 
-RTCDtlsTransport::RTCDtlsTransport(
-    ExecutionContext* context,
-    rtc::scoped_refptr<webrtc::DtlsTransportInterface> native_transport)
-    : ContextClient(context), native_transport_(native_transport) {}
+RTCDtlsTransport* RTCDtlsTransport::Create(ExecutionContext* context) {
+  return MakeGarbageCollected<RTCDtlsTransport>(context);
+}
+
+RTCDtlsTransport::RTCDtlsTransport(ExecutionContext* context)
+    : ContextClient(context) {}
 
 RTCDtlsTransport::~RTCDtlsTransport() {}
 
@@ -54,10 +55,6 @@ RTCIceTransport* RTCDtlsTransport::iceTransport() const {
   return nullptr;
 }
 
-webrtc::DtlsTransportInterface* RTCDtlsTransport::native_transport() {
-  return native_transport_.get();
-}
-
 const AtomicString& RTCDtlsTransport::InterfaceName() const {
   return event_target_names::kRTCDtlsTransport;
 }

third_party/blink/renderer/modules/peerconnection/rtc_dtls_transport.h

@@ -8,11 +8,6 @@
 #include "third_party/blink/renderer/bindings/core/v8/active_script_wrappable.h"
 #include "third_party/blink/renderer/core/dom/context_lifecycle_observer.h"
 #include "third_party/blink/renderer/modules/event_target_modules.h"
-#include "third_party/webrtc/api/scoped_refptr.h"
-
-namespace webrtc {
-class DtlsTransportInterface;
-}
 
 namespace blink {
 
@@ -37,9 +32,9 @@ class MODULES_EXPORT RTCDtlsTransport final : public EventTargetWithInlineData,
   USING_GARBAGE_COLLECTED_MIXIN(RTCDtlsTransport);
 
  public:
-  RTCDtlsTransport(
-      ExecutionContext* context,
-      rtc::scoped_refptr<webrtc::DtlsTransportInterface> native_context);
+  static RTCDtlsTransport* Create(ExecutionContext* context);
+
+  explicit RTCDtlsTransport(ExecutionContext* context);
   ~RTCDtlsTransport() override;
 
   // rtc_dtls_transport.idl
@@ -55,11 +50,9 @@ class MODULES_EXPORT RTCDtlsTransport final : public EventTargetWithInlineData,
   ExecutionContext* GetExecutionContext() const override;
   // For garbage collection.
   void Trace(blink::Visitor* visitor) override;
-  webrtc::DtlsTransportInterface* native_transport();
 
  private:
   HeapVector<Member<DOMArrayBuffer>> remote_certificates_;
-  rtc::scoped_refptr<webrtc::DtlsTransportInterface> native_transport_;
 };
 
 }  // namespace blink

third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.cc

@@ -87,7 +87,6 @@
 #include "third_party/blink/renderer/modules/peerconnection/rtc_data_channel.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_data_channel_event.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_data_channel_init.h"
-#include "third_party/blink/renderer/modules/peerconnection/rtc_dtls_transport.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_dtmf_sender.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_error_util.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_ice_server.h"
@@ -118,7 +117,6 @@
 #include "third_party/blink/renderer/platform/runtime_enabled_features.h"
 #include "third_party/blink/renderer/platform/wtf/std_lib_extras.h"
 #include "third_party/blink/renderer/platform/wtf/time.h"
-#include "third_party/webrtc/api/dtlstransportinterface.h"
 #include "third_party/webrtc/api/jsep.h"
 #include "third_party/webrtc/api/peerconnectioninterface.h"
 #include "third_party/webrtc/pc/sessiondescription.h"
@@ -1138,47 +1136,6 @@ bool RTCPeerConnection::HasDocumentMedia() const {
          user_media_controller->HasRequestedUserMedia();
 }
 
-RTCDtlsTransport* RTCPeerConnection::LookupDtlsTransportByMid(String mid) {
-  if (mid.IsNull())
-    return nullptr;
-  webrtc::PeerConnectionInterface* native_pc =
-      peer_handler_->NativePeerConnection();
-  if (!native_pc)
-    return nullptr;
-  auto native_transport =
-      native_pc->LookupDtlsTransportByMid(std::string(mid.Utf8().data()));
-  if (!native_transport)
-    return nullptr;
-  // Check for previously created RTCDtlsTransport objects referencing
-  // this transport.
-  auto transport_iterator = dtls_transports_by_mid_.find(mid);
-  if (transport_iterator != dtls_transports_by_mid_.end()) {
-    if (transport_iterator->value->native_transport() !=
-        native_transport.get()) {
-      // The mid's transport has changed. Erase the reference to
-      // the old transport, and continue.
-      dtls_transports_by_mid_.erase(transport_iterator);
-    } else {
-      return transport_iterator->value;
-    }
-  }
-
-  // Check if the same transport has been returned for another mid.
-  for (auto const& transport_iterator : dtls_transports_by_mid_) {
-    if (transport_iterator.value->native_transport() ==
-        native_transport.get()) {
-      dtls_transports_by_mid_.insert(mid, transport_iterator.value);
-      return transport_iterator.value;
-    }
-  }
-  // The transport is previously unseen. Create object and hold on to
-  // a reference to it.
-  RTCDtlsTransport* transport = MakeGarbageCollected<RTCDtlsTransport>(
-      GetExecutionContext(), native_transport);
-  dtls_transports_by_mid_.insert(mid, transport);
-  return transport;
-}
-
 void RTCPeerConnection::ReportSetSdpUsage(
     SetSdpOperationType operation_type,
     const RTCSessionDescriptionInit* session_description_init) const {
@@ -2380,8 +2337,8 @@ RTCRtpReceiver* RTCPeerConnection::CreateOrUpdateReceiver(
   RTCRtpReceiver* receiver;
   if (receiver_it == rtp_receivers_.end()) {
     // Create new receiver.
-    receiver = MakeGarbageCollected<RTCRtpReceiver>(
-        this, std::move(web_receiver), track, MediaStreamVector());
+    receiver = MakeGarbageCollected<RTCRtpReceiver>(std::move(web_receiver),
+                                                    track, MediaStreamVector());
     // Receiving tracks should be muted by default. SetReadyState() propagates
     // the related state changes to ensure it is muted on all layers. It also
     // fires events - which is not desired - but because they fire synchronously
@@ -2617,7 +2574,7 @@ void RTCPeerConnection::DidAddReceiverPlanB(
   }
   DCHECK(FindReceiver(*web_receiver) == rtp_receivers_.end());
   RTCRtpReceiver* rtp_receiver = MakeGarbageCollected<RTCRtpReceiver>(
-      this, std::move(web_receiver), track, streams);
+      std::move(web_receiver), track, streams);
   rtp_receivers_.push_back(rtp_receiver);
   ScheduleDispatchEvent(MakeGarbageCollected<RTCTrackEvent>(
       rtp_receiver, rtp_receiver->track(), streams, nullptr));
@@ -3054,7 +3011,6 @@ void RTCPeerConnection::Trace(blink::Visitor* visitor) {
   visitor->Trace(transceivers_);
   visitor->Trace(dispatch_scheduled_event_runner_);
   visitor->Trace(scheduled_events_);
-  visitor->Trace(dtls_transports_by_mid_);
   EventTargetWithInlineData::Trace(visitor);
   PausableObject::Trace(visitor);
   MediaStreamObserver::Trace(visitor);

third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.h

@@ -62,7 +62,6 @@ class MediaStreamTrack;
 class MediaStreamTrackOrString;
 class RTCAnswerOptions;
 class RTCConfiguration;
-class RTCDtlsTransport;
 class RTCDTMFSender;
 class RTCDataChannel;
 class RTCDataChannelInit;
@@ -335,8 +334,6 @@ class MODULES_EXPORT RTCPeerConnection final
   // getUserMedia().
   bool HasDocumentMedia() const;
 
-  // Look up, and potentially create, a DTLSTransport object.
-  RTCDtlsTransport* LookupDtlsTransportByMid(String mid);
   void Trace(blink::Visitor*) override;
 
  private:
@@ -497,11 +494,6 @@ class MODULES_EXPORT RTCPeerConnection final
   HeapVector<Member<RTCRtpReceiver>> rtp_receivers_;
   HeapVector<Member<RTCRtpTransceiver>> transceivers_;
 
-  // A map of all transports that have been looked up by MID.
-  // A transport may be referenced by more than one mid, so may
-  // be present multiple times in the table.
-  HeapHashMap<String, Member<RTCDtlsTransport>> dtls_transports_by_mid_;
-
   std::unique_ptr<WebRTCPeerConnectionHandler> peer_handler_;
 
   Member<AsyncMethodRunner<RTCPeerConnection>> dispatch_scheduled_event_runner_;

third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc

@@ -7,7 +7,6 @@
 #include "third_party/blink/public/platform/web_media_stream.h"
 #include "third_party/blink/public/platform/web_media_stream_track.h"
 #include "third_party/blink/public/platform/web_rtc_rtp_contributing_source.h"
-#include "third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_rtp_capabilities.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_stats_report.h"
 #include "third_party/blink/renderer/modules/peerconnection/web_rtc_stats_report_callback_resolver.h"
@@ -18,12 +17,10 @@
 
 namespace blink {
 
-RTCRtpReceiver::RTCRtpReceiver(RTCPeerConnection* pc,
-                               std::unique_ptr<WebRTCRtpReceiver> receiver,
+RTCRtpReceiver::RTCRtpReceiver(std::unique_ptr<WebRTCRtpReceiver> receiver,
                                MediaStreamTrack* track,
                                MediaStreamVector streams)
-    : pc_(pc),
-      receiver_(std::move(receiver)),
+    : receiver_(std::move(receiver)),
       track_(track),
       streams_(std::move(streams)) {
   DCHECK(receiver_);
@@ -34,17 +31,6 @@ MediaStreamTrack* RTCRtpReceiver::track() const {
   return track_;
 }
 
-RTCDtlsTransport* RTCRtpReceiver::transport() {
-  if (!transceiver_)
-    return nullptr;
-  return pc_->LookupDtlsTransportByMid(transceiver_->mid());
-}
-
-RTCDtlsTransport* RTCRtpReceiver::rtcp_transport() {
-  // Chrome does not support turning off RTCP-mux.
-  return nullptr;
-}
-
 const HeapVector<Member<RTCRtpContributingSource>>&
 RTCRtpReceiver::getContributingSources() {
   UpdateSourcesIfNeeded();
@@ -71,10 +57,6 @@ void RTCRtpReceiver::set_streams(MediaStreamVector streams) {
   streams_ = std::move(streams);
 }
 
-void RTCRtpReceiver::set_transceiver(RTCRtpTransceiver* transceiver) {
-  transceiver_ = transceiver;
-}
-
 void RTCRtpReceiver::UpdateSourcesIfNeeded() {
   if (!contributing_sources_needs_updating_)
     return;
@@ -110,11 +92,9 @@ void RTCRtpReceiver::SetContributingSourcesNeedsUpdating() {
 }
 
 void RTCRtpReceiver::Trace(blink::Visitor* visitor) {
-  visitor->Trace(pc_);
   visitor->Trace(track_);
   visitor->Trace(streams_);
   visitor->Trace(contributing_sources_);
-  visitor->Trace(transceiver_);
   ScriptWrappable::Trace(visitor);
 }
 

third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.h

@@ -18,10 +18,7 @@
 #include "third_party/blink/renderer/platform/heap/visitor.h"
 
 namespace blink {
-class RTCDtlsTransport;
-class RTCPeerConnection;
 class RTCRtpCapabilities;
-class RTCRtpTransceiver;
 
 // https://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
 class RTCRtpReceiver final : public ScriptWrappable {
@@ -29,29 +26,24 @@ class RTCRtpReceiver final : public ScriptWrappable {
 
  public:
   // Takes ownership of the receiver.
-  RTCRtpReceiver(RTCPeerConnection*,
-                 std::unique_ptr<WebRTCRtpReceiver>,
+  RTCRtpReceiver(std::unique_ptr<WebRTCRtpReceiver>,
                  MediaStreamTrack*,
                  MediaStreamVector);
 
   static RTCRtpCapabilities* getCapabilities(const String& kind);
 
   MediaStreamTrack* track() const;
-  RTCDtlsTransport* transport();
-  RTCDtlsTransport* rtcp_transport();
   const HeapVector<Member<RTCRtpContributingSource>>& getContributingSources();
   ScriptPromise getStats(ScriptState*);
 
   const WebRTCRtpReceiver& web_receiver() const;
   MediaStreamVector streams() const;
   void set_streams(MediaStreamVector streams);
-  void set_transceiver(RTCRtpTransceiver*);
   void UpdateSourcesIfNeeded();
 
   void Trace(blink::Visitor*) override;
 
  private:
-  Member<RTCPeerConnection> pc_;
   void SetContributingSourcesNeedsUpdating();
 
   std::unique_ptr<WebRTCRtpReceiver> receiver_;
@@ -61,7 +53,6 @@ class RTCRtpReceiver final : public ScriptWrappable {
   // The current contributing sources (|getContributingSources|).
   HeapVector<Member<RTCRtpContributingSource>> contributing_sources_;
   bool contributing_sources_needs_updating_ = true;
-  Member<RTCRtpTransceiver> transceiver_;
 };
 
 }  // namespace blink

third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.idl

@@ -6,8 +6,6 @@
 [Exposed=Window]
 interface RTCRtpReceiver {
     readonly attribute MediaStreamTrack track;
-    [RuntimeEnabled=RTCDtlsTransport] readonly attribute RTCDtlsTransport? transport;
-    [RuntimeEnabled=RTCDtlsTransport] readonly attribute RTCDtlsTransport? rtcp_transport;
     static RTCRtpCapabilities?             getCapabilities(DOMString kind);
     sequence<RTCRtpContributingSource> getContributingSources();
     [CallWith=ScriptState] Promise<RTCStatsReport> getStats();

third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc

@@ -274,17 +274,6 @@ MediaStreamTrack* RTCRtpSender::track() {
   return track_;
 }
 
-RTCDtlsTransport* RTCRtpSender::transport() {
-  if (!transceiver_)
-    return nullptr;
-  return pc_->LookupDtlsTransportByMid(transceiver_->mid());
-}
-
-RTCDtlsTransport* RTCRtpSender::rtcp_transport() {
-  // Chrome does not support turning off RTCP-mux.
-  return nullptr;
-}
-
 ScriptPromise RTCRtpSender::replaceTrack(ScriptState* script_state,
                                          MediaStreamTrack* with_track) {
   ScriptPromiseResolver* resolver = ScriptPromiseResolver::Create(script_state);
@@ -452,10 +441,6 @@ void RTCRtpSender::set_streams(MediaStreamVector streams) {
   streams_ = std::move(streams);
 }
 
-void RTCRtpSender::set_transceiver(RTCRtpTransceiver* transceiver) {
-  transceiver_ = transceiver;
-}
-
 RTCDTMFSender* RTCRtpSender::dtmf() {
   // Lazy initialization of dtmf_ to avoid overhead when not used.
   if (!dtmf_ && kind_ == "audio") {
@@ -476,7 +461,6 @@ void RTCRtpSender::Trace(blink::Visitor* visitor) {
   visitor->Trace(dtmf_);
   visitor->Trace(streams_);
   visitor->Trace(last_returned_parameters_);
-  visitor->Trace(transceiver_);
   ScriptWrappable::Trace(visitor);
 }
 

third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.h

@@ -20,11 +20,9 @@
 namespace blink {
 
 class MediaStreamTrack;
-class RTCDtlsTransport;
 class RTCDTMFSender;
 class RTCPeerConnection;
 class RTCRtpCapabilities;
-class RTCRtpTransceiver;
 
 webrtc::RtpEncodingParameters ToRtpEncodingParameters(
     const RTCRtpEncodingParameters*);
@@ -43,8 +41,6 @@ class RTCRtpSender final : public ScriptWrappable {
                MediaStreamVector streams);
 
   MediaStreamTrack* track();
-  RTCDtlsTransport* transport();
-  RTCDtlsTransport* rtcp_transport();
   ScriptPromise replaceTrack(ScriptState*, MediaStreamTrack*);
   RTCDTMFSender* dtmf();
   static RTCRtpCapabilities* getCapabilities(const String& kind);
@@ -59,7 +55,6 @@ class RTCRtpSender final : public ScriptWrappable {
   void ClearLastReturnedParameters();
   MediaStreamVector streams() const;
   void set_streams(MediaStreamVector streams);
-  void set_transceiver(RTCRtpTransceiver*);
 
   void Trace(blink::Visitor*) override;
 
@@ -73,7 +68,6 @@ class RTCRtpSender final : public ScriptWrappable {
   Member<RTCDTMFSender> dtmf_;
   MediaStreamVector streams_;
   Member<RTCRtpSendParameters> last_returned_parameters_;
-  Member<RTCRtpTransceiver> transceiver_;
 };
 
 }  // namespace blink

third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.idl

@@ -6,8 +6,6 @@
 [Exposed=Window]
 interface RTCRtpSender {
     readonly attribute MediaStreamTrack? track;
-    [RuntimeEnabled=RTCDtlsTransport] readonly attribute RTCDtlsTransport? transport;
-    [RuntimeEnabled=RTCDtlsTransport] readonly attribute RTCDtlsTransport? rtcp_transport;
     static RTCRtpCapabilities? getCapabilities(DOMString kind);
     [RuntimeEnabled=RTCRtpSenderParameters, CallWith=ScriptState] Promise<void> setParameters(optional RTCRtpSendParameters parameters);
     [RuntimeEnabled=RTCRtpSenderParameters] RTCRtpSendParameters getParameters();

third_party/blink/renderer/modules/peerconnection/rtc_rtp_transceiver.cc

@@ -100,8 +100,6 @@ RTCRtpTransceiver::RTCRtpTransceiver(
   DCHECK(sender_);
   DCHECK(receiver_);
   UpdateMembers();
-  sender_->set_transceiver(this);
-  receiver_->set_transceiver(this);
 }
 
 String RTCRtpTransceiver::mid() const {

third_party/blink/renderer/platform/testing/testing_platform_support_with_web_rtc.cc

@@ -216,11 +216,6 @@ WebString MockWebRTCPeerConnectionHandler::Id() const {
   return WebString();
 }
 
-webrtc::PeerConnectionInterface*
-MockWebRTCPeerConnectionHandler::NativePeerConnection() {
-  return nullptr;
-}
-
 std::unique_ptr<WebRTCPeerConnectionHandler>
 TestingPlatformSupportWithWebRTC::CreateRTCPeerConnectionHandler(
     WebRTCPeerConnectionHandlerClient*,

third_party/blink/renderer/platform/testing/testing_platform_support_with_web_rtc.h

@@ -64,7 +64,6 @@ class MockWebRTCPeerConnectionHandler : public WebRTCPeerConnectionHandler {
       const WebRTCDataChannelInit&) override;
   void Stop() override;
   WebString Id() const override;
-  webrtc::PeerConnectionInterface* NativePeerConnection() override;
 };
 
 class TestingPlatformSupportWithWebRTC : public TestingPlatformSupport {

third_party/blink/web_tests/external/wpt/webrtc/RTCRtpSender-transport.https-expected.txt

-This is a testharness.js-based test.
-PASS RTCRtpSender.transport is null when unconnected
-FAIL RTCRtpSender/receiver.transport has a value when connected assert_not_equals: got disallowed value null
-FAIL RTCRtpSender/receiver.transport at the right time, with bundle policy balanced assert_not_equals: got disallowed value null
-FAIL RTCRtpSender/receiver.transport at the right time, with bundle policy max-bundle assert_not_equals: got disallowed value null
-FAIL RTCRtpSender/receiver.transport at the right time, with bundle policy max-compat assert_not_equals: got disallowed value null
-Harness: the test ran to completion.
-

third_party/blink/web_tests/external/wpt/webrtc/RTCRtpSender-transport.https.html

-<!doctype html>
-<meta charset=utf-8>
-<title>RTCRtpSender.transport</title>
-<script src="/resources/testharness.js"></script>
-<script src="/resources/testharnessreport.js"></script>
-<script src="dictionary-helper.js"></script>
-<script src="RTCPeerConnection-helper.js"></script>
-<script>
-  'use strict';
-
-  // Spec link: http://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport
-  promise_test(async t => {
-    const caller = new RTCPeerConnection();
-    t.add_cleanup(() => caller.close());
-    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
-    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
-    const [track] = stream.getTracks();
-    const sender = caller.addTrack(track);
-    assert_equals(sender.transport, null);
-  }, 'RTCRtpSender.transport is null when unconnected');
-
-  // Test for the simple/happy path of connecting a single track
-  promise_test(async t => {
-    const caller = new RTCPeerConnection();
-    t.add_cleanup(() => caller.close());
-    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
-    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
-    const [track] = stream.getTracks();
-    const sender = caller.addTrack(track);
-    const callee = new RTCPeerConnection();
-    t.add_cleanup(() => callee.close());
-    exchangeIceCandidates(caller, callee);
-    await exchangeOfferAndListenToOntrack(t, caller, callee);
-    assert_not_equals(sender.transport, null);
-    const [transceiver] = caller.getTransceivers();
-    assert_equals(transceiver.sender.transport,
-                  transceiver.receiver.transport);
-  }, 'RTCRtpSender/receiver.transport has a value when connected');
-
-  // Test with multiple tracks, and checking details of when things show up
-  // for different bundle policies.
-  for (let bundle_policy of ['balanced', 'max-bundle', 'max-compat']) {
-    promise_test(async t => {
-        const caller = new RTCPeerConnection({bundlePolicy: bundle_policy});
-      t.add_cleanup(() => caller.close());
-      const stream = await navigator.mediaDevices.getUserMedia(
-          {audio: true, video:true});
-      t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
-      const [track1, track2] = stream.getTracks();
-      const sender1 = caller.addTrack(track1);
-      const sender2 = caller.addTrack(track2);
-      const callee = new RTCPeerConnection();
-      t.add_cleanup(() => callee.close());
-      exchangeIceCandidates(caller, callee);
-      const offer = await caller.createOffer();
-      assert_equals(sender1.transport, null);
-      assert_equals(sender2.transport, null);
-      await caller.setLocalDescription(offer);
-      assert_not_equals(sender1.transport, null);
-      assert_not_equals(sender2.transport, null);
-      const [caller_transceiver1, caller_transceiver2] = caller.getTransceivers();
-      assert_equals(sender1.transport, caller_transceiver1.sender.transport);
-      if (bundle_policy == 'max-bundle') {
-        assert_equals(caller_transceiver1.sender.transport,
-                      caller_transceiver2.sender.transport);
-      } else {
-        assert_not_equals(caller_transceiver1.sender.transport,
-                          caller_transceiver2.sender.transport);
-      }
-      await callee.setRemoteDescription(offer);
-      const [callee_transceiver1, callee_transceiver2] = callee.getTransceivers();
-      assert_not_equals(callee_transceiver1.receiver.transport, null);
-      assert_not_equals(callee_transceiver2.receiver.transport, null);
-      const answer = await callee.createAnswer();
-      await callee.setLocalDescription(answer);
-      // At this point, bundle should have kicked in.
-      assert_equals(callee_transceiver1.receiver.transport,
-                    callee_transceiver2.receiver.transport);
-      await caller.setRemoteDescription(answer);
-      assert_equals(caller_transceiver1.receiver.transport,
-                    caller_transceiver2.receiver.transport);
-    }, 'RTCRtpSender/receiver.transport at the right time, with bundle policy ' + bundle_policy);
-  }
- </script>

third_party/blink/web_tests/external/wpt/webrtc/idlharness.https.window-expected.txt

@@ -228,7 +228,7 @@ PASS RTCRtpSender interface: existence and properties of interface prototype obj
 PASS RTCRtpSender interface: existence and properties of interface prototype object's "constructor" property
 PASS RTCRtpSender interface: existence and properties of interface prototype object's @@unscopables property
 PASS RTCRtpSender interface: attribute track
-PASS RTCRtpSender interface: attribute transport
+FAIL RTCRtpSender interface: attribute transport assert_true: The prototype object must have a property "transport" expected true got false
 FAIL RTCRtpSender interface: attribute rtcpTransport assert_true: The prototype object must have a property "rtcpTransport" expected true got false
 PASS RTCRtpSender interface: operation getCapabilities(DOMString)
 FAIL RTCRtpSender interface: operation setParameters(RTCRtpSendParameters) assert_equals: property has wrong .length expected 1 but got 0
@@ -260,7 +260,7 @@ PASS RTCRtpReceiver interface: existence and properties of interface prototype o
 PASS RTCRtpReceiver interface: existence and properties of interface prototype object's "constructor" property
 PASS RTCRtpReceiver interface: existence and properties of interface prototype object's @@unscopables property
 PASS RTCRtpReceiver interface: attribute track
-PASS RTCRtpReceiver interface: attribute transport
+FAIL RTCRtpReceiver interface: attribute transport assert_true: The prototype object must have a property "transport" expected true got false
 FAIL RTCRtpReceiver interface: attribute rtcpTransport assert_true: The prototype object must have a property "rtcpTransport" expected true got false
 PASS RTCRtpReceiver interface: operation getCapabilities(DOMString)
 FAIL RTCRtpReceiver interface: operation getParameters() assert_own_property: interface prototype object missing non-static operation expected property "getParameters" missing

third_party/blink/web_tests/virtual/webrtc-wpt-unified-plan/external/wpt/webrtc/RTCRtpSender-transport.https-expected.txt

-This is a testharness.js-based test.
-PASS RTCRtpSender.transport is null when unconnected
-PASS RTCRtpSender/receiver.transport has a value when connected
-PASS RTCRtpSender/receiver.transport at the right time, with bundle policy balanced
-PASS RTCRtpSender/receiver.transport at the right time, with bundle policy max-bundle
-PASS RTCRtpSender/receiver.transport at the right time, with bundle policy max-compat
-Harness: the test ran to completion.
-

third_party/blink/web_tests/virtual/webrtc-wpt-unified-plan/external/wpt/webrtc/idlharness.https.window-expected.txt

@@ -228,7 +228,7 @@ PASS RTCRtpSender interface: existence and properties of interface prototype obj
 PASS RTCRtpSender interface: existence and properties of interface prototype object's "constructor" property
 PASS RTCRtpSender interface: existence and properties of interface prototype object's @@unscopables property
 PASS RTCRtpSender interface: attribute track
-PASS RTCRtpSender interface: attribute transport
+FAIL RTCRtpSender interface: attribute transport assert_true: The prototype object must have a property "transport" expected true got false
 FAIL RTCRtpSender interface: attribute rtcpTransport assert_true: The prototype object must have a property "rtcpTransport" expected true got false
 PASS RTCRtpSender interface: operation getCapabilities(DOMString)
 FAIL RTCRtpSender interface: operation setParameters(RTCRtpSendParameters) assert_equals: property has wrong .length expected 1 but got 0
@@ -240,7 +240,7 @@ PASS RTCRtpSender interface: attribute dtmf
 PASS RTCRtpSender must be primary interface of new RTCPeerConnection().addTransceiver('audio').sender
 PASS Stringification of new RTCPeerConnection().addTransceiver('audio').sender
 PASS RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "track" with the proper type
-PASS RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "transport" with the proper type
+FAIL RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "transport" with the proper type assert_inherits: property "transport" not found in prototype chain
 FAIL RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "rtcpTransport" with the proper type assert_inherits: property "rtcpTransport" not found in prototype chain
 PASS RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "getCapabilities(DOMString)" with the proper type
 PASS RTCRtpSender interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError
@@ -260,7 +260,7 @@ PASS RTCRtpReceiver interface: existence and properties of interface prototype o
 PASS RTCRtpReceiver interface: existence and properties of interface prototype object's "constructor" property
 PASS RTCRtpReceiver interface: existence and properties of interface prototype object's @@unscopables property
 PASS RTCRtpReceiver interface: attribute track
-PASS RTCRtpReceiver interface: attribute transport
+FAIL RTCRtpReceiver interface: attribute transport assert_true: The prototype object must have a property "transport" expected true got false
 FAIL RTCRtpReceiver interface: attribute rtcpTransport assert_true: The prototype object must have a property "rtcpTransport" expected true got false
 PASS RTCRtpReceiver interface: operation getCapabilities(DOMString)
 FAIL RTCRtpReceiver interface: operation getParameters() assert_own_property: interface prototype object missing non-static operation expected property "getParameters" missing
@@ -270,7 +270,7 @@ PASS RTCRtpReceiver interface: operation getStats()
 PASS RTCRtpReceiver must be primary interface of new RTCPeerConnection().addTransceiver('audio').receiver
 PASS Stringification of new RTCPeerConnection().addTransceiver('audio').receiver
 PASS RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "track" with the proper type
-PASS RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "transport" with the proper type
+FAIL RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "transport" with the proper type assert_inherits: property "transport" not found in prototype chain
 FAIL RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "rtcpTransport" with the proper type assert_inherits: property "rtcpTransport" not found in prototype chain
 PASS RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getCapabilities(DOMString)" with the proper type
 PASS RTCRtpReceiver interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').receiver with too few arguments must throw TypeError

third_party/blink/web_tests/webexposed/global-interface-listing-expected.txt

@@ -5674,9 +5674,7 @@ interface RTCRtpContributingSource
 interface RTCRtpReceiver
     static method getCapabilities
     attribute @@toStringTag
-    getter rtcp_transport
     getter track
-    getter transport
     method constructor
     method getContributingSources
     method getStats
@@ -5684,9 +5682,7 @@ interface RTCRtpSender
     static method getCapabilities
     attribute @@toStringTag
     getter dtmf
-    getter rtcp_transport
     getter track
-    getter transport
     method constructor
     method getParameters
     method getStats