Implement RTCRtpReceiver.getParameters()

Intent: https://groups.google.com/a/chromium.org/d/msg/blink-dev/LXJUzctTVfM/p4J-WrzbBgAJ

Bug: 908911
Change-Id: I8e82cc588fb991458e2372fe1dcb9db459cd36b5
Reviewed-on: https://chromium-review.googlesource.com/c/1363276Reviewed-by: Henrik Boström <hbos@chromium.org>
Reviewed-by: Philip Jägenstedt <foolip@chromium.org>
Commit-Queue: Florent Castelli <orphis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#615518}

content/renderer/media/webrtc/fake_rtc_rtp_transceiver.cc

@@ -129,6 +129,12 @@ void FakeRTCRtpReceiver::GetStats(
   NOTIMPLEMENTED();
 }
 
+std::unique_ptr<webrtc::RtpParameters> FakeRTCRtpReceiver::GetParameters()
+    const {
+  NOTIMPLEMENTED();
+  return nullptr;
+}
+
 FakeRTCRtpTransceiver::FakeRTCRtpTransceiver(
     base::Optional<std::string> mid,
     FakeRTCRtpSender sender,

content/renderer/media/webrtc/fake_rtc_rtp_transceiver.h

@@ -67,6 +67,7 @@ class CONTENT_EXPORT FakeRTCRtpReceiver : public blink::WebRTCRtpReceiver {
   GetSources() override;
   void GetStats(std::unique_ptr<blink::WebRTCStatsReportCallback>,
                 blink::RTCStatsFilter) override;
+  std::unique_ptr<webrtc::RtpParameters> GetParameters() const override;
 
  private:
   blink::WebMediaStreamTrack track_;

content/renderer/media/webrtc/rtc_rtp_receiver.cc

@@ -154,6 +154,11 @@ class RTCRtpReceiver::RTCRtpReceiverInternal
                        std::move(callback), filter));
   }
 
+  std::unique_ptr<webrtc::RtpParameters> GetParameters() {
+    return std::make_unique<webrtc::RtpParameters>(
+        webrtc_receiver_->GetParameters());
+  }
+
  private:
   friend struct RTCRtpReceiver::RTCRtpReceiverInternalTraits;
 
@@ -260,6 +265,10 @@ void RTCRtpReceiver::GetStats(
   internal_->GetStats(std::move(callback), filter);
 }
 
+std::unique_ptr<webrtc::RtpParameters> RTCRtpReceiver::GetParameters() const {
+  return internal_->GetParameters();
+}
+
 RTCRtpReceiverOnlyTransceiver::RTCRtpReceiverOnlyTransceiver(
     std::unique_ptr<blink::WebRTCRtpReceiver> receiver)
     : receiver_(std::move(receiver)) {

content/renderer/media/webrtc/rtc_rtp_receiver.h

@@ -119,6 +119,7 @@ class CONTENT_EXPORT RTCRtpReceiver : public blink::WebRTCRtpReceiver {
   GetSources() override;
   void GetStats(std::unique_ptr<blink::WebRTCStatsReportCallback>,
                 blink::RTCStatsFilter) override;
+  std::unique_ptr<webrtc::RtpParameters> GetParameters() const override;
 
  private:
   class RTCRtpReceiverInternal;

third_party/blink/public/platform/web_rtc_rtp_receiver.h

@@ -10,6 +10,7 @@
 #include "third_party/blink/public/platform/web_rtc_stats.h"
 #include "third_party/blink/public/platform/web_string.h"
 #include "third_party/blink/public/platform/web_vector.h"
+#include "third_party/webrtc/api/rtpparameters.h"
 
 namespace blink {
 
@@ -34,6 +35,7 @@ class BLINK_PLATFORM_EXPORT WebRTCRtpReceiver {
   GetSources() = 0;
   virtual void GetStats(std::unique_ptr<blink::WebRTCStatsReportCallback>,
                         RTCStatsFilter) = 0;
+  virtual std::unique_ptr<webrtc::RtpParameters> GetParameters() const = 0;
 };
 
 }  // namespace blink

third_party/blink/renderer/modules/modules_idl_files.gni

@@ -637,10 +637,12 @@ modules_dictionary_idl_files =
           "peerconnection/rtc_rtp_codec_capability.idl",
           "peerconnection/rtc_rtp_codec_parameters.idl",
           "peerconnection/rtc_rtp_coding_parameters.idl",
+          "peerconnection/rtc_rtp_decoding_parameters.idl",
           "peerconnection/rtc_rtp_encoding_parameters.idl",
           "peerconnection/rtc_rtp_header_extension_capability.idl",
           "peerconnection/rtc_rtp_header_extension_parameters.idl",
           "peerconnection/rtc_rtp_parameters.idl",
+          "peerconnection/rtc_rtp_receive_parameters.idl",
           "peerconnection/rtc_rtp_send_parameters.idl",
           "peerconnection/rtc_rtp_transceiver_init.idl",
           "peerconnection/rtc_session_description_init.idl",

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

+// Copyright 2018 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// https://w3c.github.io/webrtc-pc/#dom-rtcrtpdecodingparameters
+dictionary RTCRtpDecodingParameters : RTCRtpCodingParameters {
+};
\ No newline at end of file

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

+// Copyright 2018 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// https://w3c.github.io/webrtc-pc/#rtcreceivertpparameters
+dictionary RTCRtpReceiveParameters : RTCRtpParameters {
+    required sequence<RTCRtpDecodingParameters> encodings;
+};
\ No newline at end of file

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

@@ -8,6 +8,7 @@
 #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_rtp_capabilities.h"
+#include "third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.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"
 #include "third_party/blink/renderer/platform/bindings/microtask.h"
@@ -147,4 +148,44 @@ RTCRtpCapabilities* RTCRtpReceiver::getCapabilities(const String& kind) {
   return capabilities;
 }
 
+RTCRtpReceiveParameters* RTCRtpReceiver::getParameters() {
+  RTCRtpReceiveParameters* parameters = RTCRtpReceiveParameters::Create();
+  std::unique_ptr<webrtc::RtpParameters> webrtc_parameters =
+      receiver_->GetParameters();
+
+  RTCRtcpParameters* rtcp = RTCRtcpParameters::Create();
+  rtcp->setReducedSize(webrtc_parameters->rtcp.reduced_size);
+  parameters->setRtcp(rtcp);
+
+  HeapVector<Member<RTCRtpDecodingParameters>> encodings;
+  encodings.ReserveCapacity(
+      SafeCast<wtf_size_t>(webrtc_parameters->encodings.size()));
+  for (const auto& webrtc_encoding : webrtc_parameters->encodings) {
+    RTCRtpDecodingParameters* encoding = RTCRtpDecodingParameters::Create();
+    if (!webrtc_encoding.rid.empty()) {
+      // TODO(orphis): Add rid when supported by WebRTC
+    }
+    encodings.push_back(encoding);
+  }
+  parameters->setEncodings(encodings);
+
+  HeapVector<Member<RTCRtpHeaderExtensionParameters>> headers;
+  headers.ReserveCapacity(
+      SafeCast<wtf_size_t>(webrtc_parameters->header_extensions.size()));
+  for (const auto& webrtc_header : webrtc_parameters->header_extensions) {
+    headers.push_back(ToRtpHeaderExtensionParameters(webrtc_header));
+  }
+  parameters->setHeaderExtensions(headers);
+
+  HeapVector<Member<RTCRtpCodecParameters>> codecs;
+  codecs.ReserveCapacity(
+      SafeCast<wtf_size_t>(webrtc_parameters->codecs.size()));
+  for (const auto& webrtc_codec : webrtc_parameters->codecs) {
+    codecs.push_back(ToRtpCodecParameters(webrtc_codec));
+  }
+  parameters->setCodecs(codecs);
+
+  return parameters;
+}
+
 }  // namespace blink

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

@@ -12,6 +12,7 @@
 #include "third_party/blink/renderer/modules/mediastream/media_stream.h"
 #include "third_party/blink/renderer/modules/mediastream/media_stream_track.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_rtp_contributing_source.h"
+#include "third_party/blink/renderer/modules/peerconnection/rtc_rtp_receive_parameters.h"
 #include "third_party/blink/renderer/platform/bindings/script_wrappable.h"
 #include "third_party/blink/renderer/platform/heap/garbage_collected.h"
 #include "third_party/blink/renderer/platform/heap/member.h"
@@ -33,6 +34,7 @@ class RTCRtpReceiver final : public ScriptWrappable {
   static RTCRtpCapabilities* getCapabilities(const String& kind);
 
   MediaStreamTrack* track() const;
+  RTCRtpReceiveParameters* getParameters();
   const HeapVector<Member<RTCRtpContributingSource>>& getContributingSources();
   ScriptPromise getStats(ScriptState*);
 

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

@@ -6,7 +6,8 @@
 [Exposed=Window]
 interface RTCRtpReceiver {
     readonly attribute MediaStreamTrack track;
-    static RTCRtpCapabilities?             getCapabilities(DOMString kind);
+    static RTCRtpCapabilities?            getCapabilities(DOMString kind);
+    RTCRtpReceiveParameters               getParameters();
     sequence<RTCRtpContributingSource> getContributingSources();
     [CallWith=ScriptState] Promise<RTCStatsReport> getStats();
     // TODO(hbos): Support every member of the spec. https://crbug.com/700916

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

@@ -255,6 +255,37 @@ webrtc::RtpEncodingParameters ToRtpEncodingParameters(
   return webrtc_encoding;
 }
 
+RTCRtpHeaderExtensionParameters* ToRtpHeaderExtensionParameters(
+    const webrtc::RtpHeaderExtensionParameters& webrtc_header) {
+  RTCRtpHeaderExtensionParameters* header =
+      RTCRtpHeaderExtensionParameters::Create();
+  header->setUri(webrtc_header.uri.c_str());
+  header->setId(webrtc_header.id);
+  header->setEncrypted(webrtc_header.encrypt);
+  return header;
+}
+
+RTCRtpCodecParameters* ToRtpCodecParameters(
+    const webrtc::RtpCodecParameters& webrtc_codec) {
+  RTCRtpCodecParameters* codec = RTCRtpCodecParameters::Create();
+  codec->setPayloadType(webrtc_codec.payload_type);
+  codec->setMimeType(WTF::String::FromUTF8(webrtc_codec.mime_type().c_str()));
+  if (webrtc_codec.clock_rate)
+    codec->setClockRate(webrtc_codec.clock_rate.value());
+  if (webrtc_codec.num_channels)
+    codec->setChannels(webrtc_codec.num_channels.value());
+  if (webrtc_codec.parameters.size()) {
+    std::string sdp_fmtp_line;
+    for (const auto& parameter : webrtc_codec.parameters) {
+      if (sdp_fmtp_line.size())
+        sdp_fmtp_line += ";";
+      sdp_fmtp_line += parameter.first + "=" + parameter.second;
+    }
+    codec->setSdpFmtpLine(sdp_fmtp_line.c_str());
+  }
+  return codec;
+}
+
 RTCRtpSender::RTCRtpSender(RTCPeerConnection* pc,
                            std::unique_ptr<WebRTCRtpSender> sender,
                            String kind,
@@ -311,17 +342,17 @@ RTCRtpSendParameters* RTCRtpSender::getParameters() {
   HeapVector<Member<RTCRtpEncodingParameters>> encodings;
   encodings.ReserveCapacity(
       SafeCast<wtf_size_t>(webrtc_parameters->encodings.size()));
-  for (const auto& web_encoding : webrtc_parameters->encodings) {
+  for (const auto& webrtc_encoding : webrtc_parameters->encodings) {
     // TODO(orphis): Forward missing fields from the WebRTC library:
     // codecPayloadType, dtx, ptime, maxFramerate, scaleResolutionDownBy, rid
     RTCRtpEncodingParameters* encoding = RTCRtpEncodingParameters::Create();
-    encoding->setActive(web_encoding.active);
-    if (web_encoding.max_bitrate_bps)
-      encoding->setMaxBitrate(web_encoding.max_bitrate_bps.value());
+    encoding->setActive(webrtc_encoding.active);
+    if (webrtc_encoding.max_bitrate_bps)
+      encoding->setMaxBitrate(webrtc_encoding.max_bitrate_bps.value());
     encoding->setPriority(
-        PriorityFromDouble(web_encoding.bitrate_priority).c_str());
+        PriorityFromDouble(webrtc_encoding.bitrate_priority).c_str());
     encoding->setNetworkPriority(
-        PriorityFromDouble(web_encoding.network_priority).c_str());
+        PriorityFromDouble(webrtc_encoding.network_priority).c_str());
     encodings.push_back(encoding);
   }
   parameters->setEncodings(encodings);
@@ -329,36 +360,16 @@ RTCRtpSendParameters* RTCRtpSender::getParameters() {
   HeapVector<Member<RTCRtpHeaderExtensionParameters>> headers;
   headers.ReserveCapacity(
       SafeCast<wtf_size_t>(webrtc_parameters->header_extensions.size()));
-  for (const auto& web_header : webrtc_parameters->header_extensions) {
-    auto* header = RTCRtpHeaderExtensionParameters::Create();
-    header->setUri(web_header.uri.c_str());
-    header->setId(web_header.id);
-    header->setEncrypted(web_header.encrypt);
-    headers.push_back(header);
+  for (const auto& webrtc_header : webrtc_parameters->header_extensions) {
+    headers.push_back(ToRtpHeaderExtensionParameters(webrtc_header));
   }
   parameters->setHeaderExtensions(headers);
 
   HeapVector<Member<RTCRtpCodecParameters>> codecs;
   codecs.ReserveCapacity(
       SafeCast<wtf_size_t>(webrtc_parameters->codecs.size()));
-  for (const auto& web_codec : webrtc_parameters->codecs) {
-    auto* codec = RTCRtpCodecParameters::Create();
-    codec->setPayloadType(web_codec.payload_type);
-    codec->setMimeType(WTF::String::FromUTF8(web_codec.mime_type().c_str()));
-    if (web_codec.clock_rate)
-      codec->setClockRate(web_codec.clock_rate.value());
-    if (web_codec.num_channels)
-      codec->setChannels(web_codec.num_channels.value());
-    if (web_codec.parameters.size()) {
-      std::string sdp_fmtp_line;
-      for (const auto& parameter : web_codec.parameters) {
-        if (sdp_fmtp_line.size())
-          sdp_fmtp_line += ";";
-        sdp_fmtp_line += parameter.first + "=" + parameter.second;
-      }
-      codec->setSdpFmtpLine(sdp_fmtp_line.c_str());
-    }
-    codecs.push_back(codec);
+  for (const auto& webrtc_codec : webrtc_parameters->codecs) {
+    codecs.push_back(ToRtpCodecParameters(webrtc_codec));
   }
   parameters->setCodecs(codecs);
 

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

@@ -26,6 +26,10 @@ class RTCRtpCapabilities;
 
 webrtc::RtpEncodingParameters ToRtpEncodingParameters(
     const RTCRtpEncodingParameters*);
+RTCRtpHeaderExtensionParameters* ToRtpHeaderExtensionParameters(
+    const webrtc::RtpHeaderExtensionParameters& headers);
+RTCRtpCodecParameters* ToRtpCodecParameters(
+    const webrtc::RtpCodecParameters& codecs);
 
 // https://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 class RTCRtpSender final : public ScriptWrappable {

third_party/blink/web_tests/external/wpt/webrtc/RTCRtpParameters-helper.js

@@ -19,6 +19,8 @@ async function doOfferAnswerExchange(t, caller) {
   const answer = await callee.createAnswer();
   await callee.setLocalDescription(answer);
   await caller.setRemoteDescription(answer);
+
+  return callee;
 }
 
 /*

third_party/blink/web_tests/external/wpt/webrtc/RTCRtpReceiver-getParameters-expected.txt

 This is a testharness.js-based test.
-FAIL RTCRtpReceiver.prototype.getParameters Failed to execute 'addTransceiver' on 'RTCPeerConnection': This operation is only supported in 'unified-plan'. 'unified-plan' will become the default behavior in the future, but it is currently experimental. To try it out, construct the RTCPeerConnection with sdpSemantics:'unified-plan' present in the RTCConfiguration argument.
+FAIL getParameters() with audio receiver promise_test: Unhandled rejection with value: object "InvalidStateError: Failed to execute 'addTransceiver' on 'RTCPeerConnection': This operation is only supported in 'unified-plan'. 'unified-plan' will become the default behavior in the future, but it is currently experimental. To try it out, construct the RTCPeerConnection with sdpSemantics:'unified-plan' present in the RTCConfiguration argument."
+FAIL getParameters() with video receiver promise_test: Unhandled rejection with value: object "InvalidStateError: Failed to execute 'addTransceiver' on 'RTCPeerConnection': This operation is only supported in 'unified-plan'. 'unified-plan' will become the default behavior in the future, but it is currently experimental. To try it out, construct the RTCPeerConnection with sdpSemantics:'unified-plan' present in the RTCConfiguration argument."
+FAIL getParameters() with simulcast video receiver promise_test: Unhandled rejection with value: object "InvalidStateError: Failed to execute 'addTransceiver' on 'RTCPeerConnection': This operation is only supported in 'unified-plan'. 'unified-plan' will become the default behavior in the future, but it is currently experimental. To try it out, construct the RTCPeerConnection with sdpSemantics:'unified-plan' present in the RTCConfiguration argument."
 Harness: the test ran to completion.
 

third_party/blink/web_tests/external/wpt/webrtc/RTCRtpReceiver-getParameters.html

@@ -33,10 +33,51 @@
 
         - transactionId and degradationPreference are left undefined.
    */
-  test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
-    const { receiver } = pc.addTransceiver('audio');
-    const param = receiver.getParameters();
+    t.add_cleanup(() => pc.close());
+    pc.addTransceiver('audio');
+    const callee = await doOfferAnswerExchange(t, pc);
+    const param = callee.getReceivers()[0].getParameters();
     validateReceiverRtpParameters(param);
-  });
+
+    assert_greater_than(param.headerExtensions.length, 0);
+    assert_greater_than(param.codecs.length, 0);
+    assert_equals(param.encodings.length, 1);
+  }, 'getParameters() with audio receiver');
+
+  promise_test(async t => {
+    const pc = new RTCPeerConnection();
+    t.add_cleanup(() => pc.close());
+    pc.addTransceiver('video');
+    const callee = await doOfferAnswerExchange(t, pc);
+    const param = callee.getReceivers()[0].getParameters();
+    validateReceiverRtpParameters(param);
+
+    assert_greater_than(param.headerExtensions.length, 0);
+    assert_greater_than(param.codecs.length, 0);
+    assert_equals(param.encodings.length, 1);
+  }, 'getParameters() with video receiver');
+
+  promise_test(async t => {
+    const pc = new RTCPeerConnection();
+    t.add_cleanup(() => pc.close());
+    pc.addTransceiver('video', {
+      sendEncodings: [
+        { rid: "rid1" },
+        { rid: "rid2" }
+      ]
+    });
+    const callee = await doOfferAnswerExchange(t, pc);
+    const param = callee.getReceivers()[0].getParameters();
+    validateReceiverRtpParameters(param);
+
+    assert_greater_than(param.headerExtensions.length, 0);
+    assert_greater_than(param.codecs.length, 0);
+    assert_equals(param.encodings.length, 2, 'layer count must match');
+    assert_equals(param.encodings[0].rid, "rid1",
+      'simulcast rids must match');
+    assert_equals(param.encodings[1].rid, "rid2",
+      'simulcast rids must match');
+  }, 'getParameters() with simulcast video receiver');
 </script>

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

@@ -263,7 +263,7 @@ PASS RTCRtpReceiver interface: attribute track
 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
+PASS RTCRtpReceiver interface: operation getParameters()
 PASS RTCRtpReceiver interface: operation getContributingSources()
 FAIL RTCRtpReceiver interface: operation getSynchronizationSources() assert_own_property: interface prototype object missing non-static operation expected property "getSynchronizationSources" missing
 PASS RTCRtpReceiver interface: operation getStats()

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

@@ -4893,6 +4893,7 @@ interface RTCRtpReceiver
     getter track
     method constructor
     method getContributingSources
+    method getParameters
     method getStats
 interface RTCRtpSender
     static method getCapabilities

third_party/blink/web_tests/virtual/webrtc-wpt-unified-plan/external/wpt/webrtc/RTCRtpReceiver-getParameters-expected.txt

 This is a testharness.js-based test.
-FAIL RTCRtpReceiver.prototype.getParameters receiver.getParameters is not a function
+PASS getParameters() with audio receiver
+PASS getParameters() with video receiver
+FAIL getParameters() with simulcast video receiver promise_test: Unhandled rejection with value: object "OperationError: Failed to execute 'addTransceiver' on 'RTCPeerConnection': Attempted to create an encoder with more than 1 encoding parameter."
 Harness: the test ran to completion.
 

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

@@ -263,7 +263,7 @@ PASS RTCRtpReceiver interface: attribute track
 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
+PASS RTCRtpReceiver interface: operation getParameters()
 PASS RTCRtpReceiver interface: operation getContributingSources()
 FAIL RTCRtpReceiver interface: operation getSynchronizationSources() assert_own_property: interface prototype object missing non-static operation expected property "getSynchronizationSources" missing
 PASS RTCRtpReceiver interface: operation getStats()
@@ -274,7 +274,7 @@ FAIL RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').r
 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
-FAIL RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getParameters()" with the proper type assert_inherits: property "getParameters" not found in prototype chain
+PASS RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getParameters()" with the proper type
 PASS RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getContributingSources()" with the proper type
 FAIL RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getSynchronizationSources()" with the proper type assert_inherits: property "getSynchronizationSources" not found in prototype chain
 PASS RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getStats()" with the proper type

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

@@ -5676,6 +5676,7 @@ interface RTCRtpReceiver
     getter track
     method constructor
     method getContributingSources
+    method getParameters
     method getStats
 interface RTCRtpSender
     static method getCapabilities