1 files changed, 619 insertions, 0 deletions
diff --git a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
new file mode 100644
index 0000000000..972b81adba
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
@@ -0,0 +1,619 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "RTCRtpReceiver.h"
+#include "transport/logging.h"
+#include "mozilla/dom/MediaStreamTrack.h"
+#include "mozilla/dom/Promise.h"
+#include "transportbridge/MediaPipeline.h"
+#include "nsPIDOMWindow.h"
+#include "PrincipalHandle.h"
+#include "nsIPrincipal.h"
+#include "mozilla/dom/Document.h"
+#include "mozilla/NullPrincipal.h"
+#include "MediaTrackGraph.h"
+#include "RemoteTrackSource.h"
+#include "libwebrtcglue/RtpRtcpConfig.h"
+#include "nsString.h"
+#include "mozilla/dom/AudioStreamTrack.h"
+#include "mozilla/dom/VideoStreamTrack.h"
+#include "MediaTransportHandler.h"
+#include "jsep/JsepTransceiver.h"
+#include "mozilla/dom/RTCRtpReceiverBinding.h"
+#include "mozilla/dom/RTCRtpSourcesBinding.h"
+#include "RTCStatsReport.h"
+#include "mozilla/Preferences.h"
+#include "TransceiverImpl.h"
+#include "libwebrtcglue/AudioConduit.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(RTCRtpReceiver, mWindow, mTrack)
+NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpReceiver)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpReceiver)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpReceiver)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+LazyLogModule gReceiverLog("RTCRtpReceiver");
+
+static PrincipalHandle GetPrincipalHandle(nsPIDOMWindowInner* aWindow,
+                                          bool aPrivacyNeeded) {
+  // Set the principal used for creating the tracks. This makes the track
+  // data (audio/video samples) accessible to the receiving page. We're
+  // only certain that privacy hasn't been requested if we're connected.
+  nsCOMPtr<nsIScriptObjectPrincipal> winPrincipal = do_QueryInterface(aWindow);
+  RefPtr<nsIPrincipal> principal = winPrincipal->GetPrincipal();
+  if (NS_WARN_IF(!principal)) {
+    principal = NullPrincipal::CreateWithoutOriginAttributes();
+  } else if (aPrivacyNeeded) {
+    principal = NullPrincipal::CreateWithInheritedAttributes(principal);
+  }
+  return MakePrincipalHandle(principal);
+}
+
+static already_AddRefed<dom::MediaStreamTrack> CreateTrack(
+    nsPIDOMWindowInner* aWindow, bool aAudio,
+    const nsCOMPtr<nsIPrincipal>& aPrincipal) {
+  MediaTrackGraph* graph = MediaTrackGraph::GetInstance(
+      aAudio ? MediaTrackGraph::AUDIO_THREAD_DRIVER
+             : MediaTrackGraph::SYSTEM_THREAD_DRIVER,
+      aWindow, MediaTrackGraph::REQUEST_DEFAULT_SAMPLE_RATE,
+      MediaTrackGraph::DEFAULT_OUTPUT_DEVICE);
+
+  RefPtr<MediaStreamTrack> track;
+  RefPtr<RemoteTrackSource> trackSource;
+  if (aAudio) {
+    RefPtr<SourceMediaTrack> source =
+        graph->CreateSourceTrack(MediaSegment::AUDIO);
+    trackSource = new RemoteTrackSource(source, aPrincipal, u"remote audio"_ns);
+    track = new AudioStreamTrack(aWindow, source, trackSource);
+  } else {
+    RefPtr<SourceMediaTrack> source =
+        graph->CreateSourceTrack(MediaSegment::VIDEO);
+    trackSource = new RemoteTrackSource(source, aPrincipal, u"remote video"_ns);
+    track = new VideoStreamTrack(aWindow, source, trackSource);
+  }
+
+  // Spec says remote tracks start out muted.
+  trackSource->SetMuted(true);
+
+  return track.forget();
+}
+
+RTCRtpReceiver::RTCRtpReceiver(nsPIDOMWindowInner* aWindow, bool aPrivacyNeeded,
+                               const std::string& aPCHandle,
+                               MediaTransportHandler* aTransportHandler,
+                               JsepTransceiver* aJsepTransceiver,
+                               nsISerialEventTarget* aMainThread,
+                               nsISerialEventTarget* aStsThread,
+                               MediaSessionConduit* aConduit,
+                               TransceiverImpl* aTransceiverImpl)
+    : mWindow(aWindow),
+      mPCHandle(aPCHandle),
+      mJsepTransceiver(aJsepTransceiver),
+      mMainThread(aMainThread),
+      mStsThread(aStsThread),
+      mTransportHandler(aTransportHandler),
+      mTransceiverImpl(aTransceiverImpl) {
+  PrincipalHandle principalHandle = GetPrincipalHandle(aWindow, aPrivacyNeeded);
+  mTrack = CreateTrack(aWindow, aConduit->type() == MediaSessionConduit::AUDIO,
+                       principalHandle.get());
+  // Until Bug 1232234 is fixed, we'll get extra RTCP BYES during renegotiation,
+  // so we'll disable muting on RTCP BYE and timeout for now.
+  if (Preferences::GetBool("media.peerconnection.mute_on_bye_or_timeout",
+                           false)) {
+    aConduit->SetRtcpEventObserver(this);
+  }
+  if (aConduit->type() == MediaSessionConduit::AUDIO) {
+    mPipeline = new MediaPipelineReceiveAudio(
+        mPCHandle, aTransportHandler, mMainThread.get(), mStsThread.get(),
+        static_cast<AudioSessionConduit*>(aConduit), mTrack, principalHandle);
+  } else {
+    mPipeline = new MediaPipelineReceiveVideo(
+        mPCHandle, aTransportHandler, mMainThread.get(), mStsThread.get(),
+        static_cast<VideoSessionConduit*>(aConduit), mTrack, principalHandle);
+  }
+}
+
+RTCRtpReceiver::~RTCRtpReceiver() { MOZ_ASSERT(!mPipeline); }
+
+JSObject* RTCRtpReceiver::WrapObject(JSContext* aCx,
+                                     JS::Handle<JSObject*> aGivenProto) {
+  return RTCRtpReceiver_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+RTCDtlsTransport* RTCRtpReceiver::GetTransport() const {
+  if (!mTransceiverImpl) {
+    return nullptr;
+  }
+  return mTransceiverImpl->GetDtlsTransport();
+}
+
+already_AddRefed<Promise> RTCRtpReceiver::GetStats() {
+  nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(mWindow);
+  ErrorResult rv;
+  RefPtr<Promise> promise = Promise::Create(global, rv);
+  if (NS_WARN_IF(rv.Failed())) {
+    rv.StealNSResult();
+    return nullptr;
+  }
+
+  // Two promises; one for RTP/RTCP stats, another for ICE stats.
+  auto promises = GetStatsInternal();
+  RTCStatsPromise::All(mMainThread, promises)
+      ->Then(
+          mMainThread, __func__,
+          [promise, window = mWindow](
+              const nsTArray<UniquePtr<RTCStatsCollection>>& aStats) {
+            RefPtr<RTCStatsReport> report(new RTCStatsReport(window));
+            for (const auto& stats : aStats) {
+              report->Incorporate(*stats);
+            }
+            promise->MaybeResolve(std::move(report));
+          },
+          [promise](nsresult aError) {
+            promise->MaybeReject(NS_ERROR_FAILURE);
+          });
+
+  return promise.forget();
+}
+
+static UniquePtr<dom::RTCStatsCollection> GetReceiverStats_s(
+    const RefPtr<MediaPipelineReceive>& aPipeline,
+    const nsString& aRecvTrackId) {
+  UniquePtr<dom::RTCStatsCollection> report(new dom::RTCStatsCollection);
+  auto asVideo = aPipeline->Conduit()->AsVideoSessionConduit();
+
+  nsString kind = asVideo.isNothing() ? u"audio"_ns : u"video"_ns;
+  nsString idstr = kind + u"_"_ns;
+  idstr.AppendInt(static_cast<uint32_t>(aPipeline->Level()));
+
+  // Add bandwidth estimation stats
+  aPipeline->Conduit()->GetBandwidthEstimation().apply([&](auto& bw) {
+    bw.mTrackIdentifier = aRecvTrackId;
+    if (!report->mBandwidthEstimations.AppendElement(bw, fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  });
+
+  Maybe<uint32_t> ssrc;
+  unsigned int ssrcval;
+  if (aPipeline->Conduit()->GetRemoteSSRC(&ssrcval)) {
+    ssrc = Some(ssrcval);
+  }
+
+  // Add frame history
+  asVideo.apply([&](const auto& conduit) {
+    if (conduit->AddFrameHistory(&report->mVideoFrameHistories)) {
+      auto& history = report->mVideoFrameHistories.LastElement();
+      history.mTrackIdentifier = aRecvTrackId;
+    }
+  });
+
+  // TODO(@@NG):ssrcs handle Conduits having multiple stats at the same level
+  // This is pending spec work
+  // Gather pipeline stats.
+  nsString localId = u"inbound_rtp_"_ns + idstr;
+  nsString remoteId;
+
+  // First, fill in remote stat with rtcp sender data, if present.
+  uint32_t packetsSent;
+  uint64_t bytesSent;
+  DOMHighResTimeStamp remoteTimestamp;
+  Maybe<DOMHighResTimeStamp> timestamp =
+      aPipeline->Conduit()->LastRtcpReceived();
+  if (timestamp.isSome() && aPipeline->Conduit()->GetRTCPSenderReport(
+                                &packetsSent, &bytesSent, &remoteTimestamp)) {
+    RTCRemoteOutboundRtpStreamStats s;
+    remoteId = u"inbound_rtcp_"_ns + idstr;
+    s.mTimestamp.Construct(*timestamp);
+    s.mId.Construct(remoteId);
+    s.mType.Construct(RTCStatsType::Remote_outbound_rtp);
+    ssrc.apply([&s](uint32_t aSsrc) { s.mSsrc.Construct(aSsrc); });
+    s.mMediaType.Construct(kind);  // mediaType is the old name for kind.
+    s.mKind.Construct(kind);
+    s.mLocalId.Construct(localId);
+    s.mPacketsSent.Construct(packetsSent);
+    s.mBytesSent.Construct(bytesSent);
+    s.mRemoteTimestamp.Construct(remoteTimestamp);
+    if (!report->mRemoteOutboundRtpStreamStats.AppendElement(s, fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  }
+
+  // Then, fill in local side (with cross-link to remote only if present)
+  RTCInboundRtpStreamStats s;
+  // TODO(bug 1496533): Should we use the time of the most-recently received
+  // RTP packet? If so, what do we use if we haven't received any RTP? Now?
+  s.mTimestamp.Construct(aPipeline->GetNow());
+  s.mId.Construct(localId);
+  s.mType.Construct(RTCStatsType::Inbound_rtp);
+  ssrc.apply([&s](uint32_t aSsrc) { s.mSsrc.Construct(aSsrc); });
+  s.mMediaType.Construct(kind);  // mediaType is the old name for kind.
+  s.mKind.Construct(kind);
+  unsigned int jitterMs, packetsLost;
+  if (aPipeline->Conduit()->GetRTPReceiverStats(&jitterMs, &packetsLost)) {
+    s.mJitter.Construct(double(jitterMs) / 1000);
+    s.mPacketsLost.Construct(packetsLost);
+  }
+  if (remoteId.Length()) {
+    s.mRemoteId.Construct(remoteId);
+  }
+  s.mPacketsReceived.Construct(aPipeline->RtpPacketsReceived());
+  s.mBytesReceived.Construct(aPipeline->RtpBytesReceived());
+
+  // Fill in packet type statistics
+  webrtc::RtcpPacketTypeCounter counters;
+  if (aPipeline->Conduit()->GetRecvPacketTypeStats(&counters)) {
+    s.mNackCount.Construct(counters.nack_packets);
+    // Fill in video only packet type stats
+    if (asVideo) {
+      s.mFirCount.Construct(counters.fir_packets);
+      s.mPliCount.Construct(counters.pli_packets);
+    }
+  }
+  // Lastly, fill in video decoder stats if this is video
+  asVideo.apply([&s](auto conduit) {
+    double framerateMean;
+    double framerateStdDev;
+    double bitrateMean;
+    double bitrateStdDev;
+    uint32_t discardedPackets;
+    uint32_t framesDecoded;
+    if (conduit->GetVideoDecoderStats(&framerateMean, &framerateStdDev,
+                                      &bitrateMean, &bitrateStdDev,
+                                      &discardedPackets, &framesDecoded)) {
+      s.mFramerateMean.Construct(framerateMean);
+      s.mFramerateStdDev.Construct(framerateStdDev);
+      s.mBitrateMean.Construct(bitrateMean);
+      s.mBitrateStdDev.Construct(bitrateStdDev);
+      s.mDiscardedPackets.Construct(discardedPackets);
+      s.mFramesDecoded.Construct(framesDecoded);
+    }
+  });
+  if (!report->mInboundRtpStreamStats.AppendElement(s, fallible)) {
+    mozalloc_handle_oom(0);
+  }
+
+  // Fill in Contributing Source statistics
+  aPipeline->GetContributingSourceStats(localId,
+                                        report->mRtpContributingSourceStats);
+
+  return report;
+}
+
+nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
+  nsTArray<RefPtr<RTCStatsPromise>> promises;
+  if (mPipeline && mHaveStartedReceiving) {
+    nsString recvTrackId;
+    MOZ_ASSERT(mTrack);
+    if (mTrack) {
+      mTrack->GetId(recvTrackId);
+    }
+    promises.AppendElement(InvokeAsync(
+        mStsThread, __func__, [pipeline = mPipeline, recvTrackId]() {
+          return RTCStatsPromise::CreateAndResolve(
+              GetReceiverStats_s(pipeline, recvTrackId), __func__);
+        }));
+
+    if (mJsepTransceiver->mTransport.mComponents) {
+      promises.AppendElement(mTransportHandler->GetIceStats(
+          mJsepTransceiver->mTransport.mTransportId, mPipeline->GetNow()));
+    }
+  }
+  return promises;
+}
+
+void RTCRtpReceiver::GetContributingSources(
+    nsTArray<RTCRtpContributingSource>& aSources) {
+  // Duplicate code...
+  if (mPipeline && mPipeline->Conduit()) {
+    RefPtr<AudioSessionConduit> conduit(
+        static_cast<AudioSessionConduit*>(mPipeline->Conduit()));
+    nsTArray<dom::RTCRtpSourceEntry> sources;
+    conduit->GetRtpSources(sources);
+    sources.RemoveElementsBy([](const dom::RTCRtpSourceEntry& aEntry) {
+      return aEntry.mSourceType != dom::RTCRtpSourceEntryType::Contributing;
+    });
+    aSources.ReplaceElementsAt(0, aSources.Length(), sources.Elements(),
+                               sources.Length());
+  }
+}
+
+void RTCRtpReceiver::GetSynchronizationSources(
+    nsTArray<dom::RTCRtpSynchronizationSource>& aSources) {
+  // Duplicate code...
+  if (mPipeline && mPipeline->Conduit()) {
+    RefPtr<AudioSessionConduit> conduit(
+        static_cast<AudioSessionConduit*>(mPipeline->Conduit()));
+    nsTArray<dom::RTCRtpSourceEntry> sources;
+    conduit->GetRtpSources(sources);
+    sources.RemoveElementsBy([](const dom::RTCRtpSourceEntry& aEntry) {
+      return aEntry.mSourceType != dom::RTCRtpSourceEntryType::Synchronization;
+    });
+    aSources.ReplaceElementsAt(0, aSources.Length(), sources.Elements(),
+                               sources.Length());
+  }
+}
+
+nsPIDOMWindowInner* RTCRtpReceiver::GetParentObject() const { return mWindow; }
+
+void RTCRtpReceiver::Shutdown() {
+  ASSERT_ON_THREAD(mMainThread);
+  if (mPipeline) {
+    mPipeline->Shutdown_m();
+    mPipeline->Conduit()->SetRtcpEventObserver(nullptr);
+    mPipeline = nullptr;
+  }
+  mTransceiverImpl = nullptr;
+}
+
+void RTCRtpReceiver::UpdateTransport() {
+  ASSERT_ON_THREAD(mMainThread);
+  if (!mHaveSetupTransport) {
+    mPipeline->SetLevel(mJsepTransceiver->GetLevel());
+    mHaveSetupTransport = true;
+  }
+
+  UniquePtr<MediaPipelineFilter> filter;
+
+  auto const& details = mJsepTransceiver->mRecvTrack.GetNegotiatedDetails();
+  if (mJsepTransceiver->HasBundleLevel() && details) {
+    std::vector<webrtc::RtpExtension> extmaps;
+    details->ForEachRTPHeaderExtension(
+        [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
+          extmaps.emplace_back(extmap.extensionname, extmap.entry);
+        });
+    filter = MakeUnique<MediaPipelineFilter>(extmaps);
+
+    // Add remote SSRCs so we can distinguish which RTP packets actually
+    // belong to this pipeline (also RTCP sender reports).
+    for (uint32_t ssrc : mJsepTransceiver->mRecvTrack.GetSsrcs()) {
+      filter->AddRemoteSSRC(ssrc);
+    }
+    for (uint32_t ssrc : mJsepTransceiver->mRecvTrack.GetRtxSsrcs()) {
+      filter->AddRemoteSSRC(ssrc);
+    }
+    auto mid = Maybe<std::string>();
+    if (GetMid() != "") {
+      mid = Some(GetMid());
+    }
+    filter->SetRemoteMediaStreamId(mid);
+
+    // Add unique payload types as a last-ditch fallback
+    auto uniquePts = mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()
+                         ->GetUniquePayloadTypes();
+    for (unsigned char& uniquePt : uniquePts) {
+      filter->AddUniquePT(uniquePt);
+    }
+  }
+
+  mPipeline->UpdateTransport_m(mJsepTransceiver->mTransport.mTransportId,
+                               std::move(filter));
+}
+
+nsresult RTCRtpReceiver::UpdateConduit() {
+  if (mPipeline->Conduit()->type() == MediaSessionConduit::VIDEO) {
+    return UpdateVideoConduit();
+  }
+  return UpdateAudioConduit();
+}
+
+nsresult RTCRtpReceiver::UpdateVideoConduit() {
+  RefPtr<VideoSessionConduit> conduit =
+      static_cast<VideoSessionConduit*>(mPipeline->Conduit());
+
+  // NOTE(pkerr) - this is new behavior. Needed because the
+  // CreateVideoReceiveStream method of the Call API will assert (in debug)
+  // and fail if a value is not provided for the remote_ssrc that will be used
+  // by the far-end sender.
+  if (!mJsepTransceiver->mRecvTrack.GetSsrcs().empty()) {
+    MOZ_LOG(gReceiverLog, LogLevel::Debug,
+            ("%s[%s]: %s Setting remote SSRC %u", mPCHandle.c_str(),
+             GetMid().c_str(), __FUNCTION__,
+             mJsepTransceiver->mRecvTrack.GetSsrcs().front()));
+    uint32_t rtxSsrc = mJsepTransceiver->mRecvTrack.GetRtxSsrcs().empty()
+                           ? 0
+                           : mJsepTransceiver->mRecvTrack.GetRtxSsrcs().front();
+    conduit->SetRemoteSSRC(mJsepTransceiver->mRecvTrack.GetSsrcs().front(),
+                           rtxSsrc);
+  }
+
+  // TODO (bug 1423041) once we pay attention to receiving MID's in RTP
+  // packets (see bug 1405495) we could make this depending on the presence of
+  // MID in the RTP packets instead of relying on the signaling.
+  if (mJsepTransceiver->HasBundleLevel() &&
+      (!mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() ||
+       !mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()->GetExt(
+           webrtc::RtpExtension::kMIdUri))) {
+    mStsThread->Dispatch(
+        NewRunnableMethod("VideoSessionConduit::DisableSsrcChanges", conduit,
+                          &VideoSessionConduit::DisableSsrcChanges));
+  }
+
+  if (mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() &&
+      mJsepTransceiver->mRecvTrack.GetActive()) {
+    const auto& details(*mJsepTransceiver->mRecvTrack.GetNegotiatedDetails());
+
+    TransceiverImpl::UpdateConduitRtpExtmap(
+        *conduit, details, MediaSessionConduitLocalDirection::kRecv);
+
+    std::vector<UniquePtr<VideoCodecConfig>> configs;
+    nsresult rv = TransceiverImpl::NegotiatedDetailsToVideoCodecConfigs(
+        details, &configs);
+
+    if (NS_FAILED(rv)) {
+      MOZ_LOG(gReceiverLog, LogLevel::Error,
+              ("%s[%s]: %s Failed to convert "
+               "JsepCodecDescriptions to VideoCodecConfigs (recv).",
+               mPCHandle.c_str(), GetMid().c_str(), __FUNCTION__));
+      return rv;
+    }
+
+    auto error =
+        conduit->ConfigureRecvMediaCodecs(configs, details.GetRtpRtcpConfig());
+
+    if (error) {
+      MOZ_LOG(gReceiverLog, LogLevel::Error,
+              ("%s[%s]: %s "
+               "ConfigureRecvMediaCodecs failed: %u",
+               mPCHandle.c_str(), GetMid().c_str(), __FUNCTION__, error));
+      return NS_ERROR_FAILURE;
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult RTCRtpReceiver::UpdateAudioConduit() {
+  RefPtr<AudioSessionConduit> conduit =
+      static_cast<AudioSessionConduit*>(mPipeline->Conduit());
+
+  if (!mJsepTransceiver->mRecvTrack.GetSsrcs().empty()) {
+    MOZ_LOG(gReceiverLog, LogLevel::Debug,
+            ("%s[%s]: %s Setting remote SSRC %u", mPCHandle.c_str(),
+             GetMid().c_str(), __FUNCTION__,
+             mJsepTransceiver->mRecvTrack.GetSsrcs().front()));
+    uint32_t rtxSsrc = mJsepTransceiver->mRecvTrack.GetRtxSsrcs().empty()
+                           ? 0
+                           : mJsepTransceiver->mRecvTrack.GetRtxSsrcs().front();
+    conduit->SetRemoteSSRC(mJsepTransceiver->mRecvTrack.GetSsrcs().front(),
+                           rtxSsrc);
+  }
+
+  if (mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() &&
+      mJsepTransceiver->mRecvTrack.GetActive()) {
+    const auto& details(*mJsepTransceiver->mRecvTrack.GetNegotiatedDetails());
+    std::vector<UniquePtr<AudioCodecConfig>> configs;
+    nsresult rv = TransceiverImpl::NegotiatedDetailsToAudioCodecConfigs(
+        details, &configs);
+
+    if (NS_FAILED(rv)) {
+      MOZ_LOG(gReceiverLog, LogLevel::Error,
+              ("%s[%s]: %s Failed to convert "
+               "JsepCodecDescriptions to AudioCodecConfigs (recv).",
+               mPCHandle.c_str(), GetMid().c_str(), __FUNCTION__));
+      return rv;
+    }
+
+    // Ensure conduit knows about extensions prior to creating streams
+    TransceiverImpl::UpdateConduitRtpExtmap(
+        *conduit, details, MediaSessionConduitLocalDirection::kRecv);
+
+    auto error = conduit->ConfigureRecvMediaCodecs(configs);
+
+    if (error) {
+      MOZ_LOG(gReceiverLog, LogLevel::Error,
+              ("%s[%s]: %s "
+               "ConfigureRecvMediaCodecs failed: %u",
+               mPCHandle.c_str(), GetMid().c_str(), __FUNCTION__, error));
+      return NS_ERROR_FAILURE;
+    }
+  }
+
+  return NS_OK;
+}
+
+void RTCRtpReceiver::Stop() {
+  if (mPipeline) {
+    mPipeline->Stop();
+  }
+}
+
+void RTCRtpReceiver::Start() {
+  mPipeline->Start();
+  mHaveStartedReceiving = true;
+}
+
+bool RTCRtpReceiver::HasTrack(const dom::MediaStreamTrack* aTrack) const {
+  return !aTrack || (mTrack == aTrack);
+}
+
+void RTCRtpReceiver::UpdateStreams(StreamAssociationChanges* aChanges) {
+  // We don't sort and use set_difference, because we need to report the
+  // added/removed streams in the order that they appear in the SDP.
+  std::set<std::string> newIds(
+      mJsepTransceiver->mRecvTrack.GetStreamIds().begin(),
+      mJsepTransceiver->mRecvTrack.GetStreamIds().end());
+  MOZ_ASSERT(mJsepTransceiver->mRecvTrack.GetRemoteSetSendBit() ||
+             newIds.empty());
+  bool needsTrackEvent = false;
+  for (const auto& id : mStreamIds) {
+    if (!newIds.count(id)) {
+      aChanges->mStreamAssociationsRemoved.push_back({mTrack, id});
+    }
+  }
+
+  std::set<std::string> oldIds(mStreamIds.begin(), mStreamIds.end());
+  for (const auto& id : mJsepTransceiver->mRecvTrack.GetStreamIds()) {
+    if (!oldIds.count(id)) {
+      needsTrackEvent = true;
+      aChanges->mStreamAssociationsAdded.push_back({mTrack, id});
+    }
+  }
+
+  mStreamIds = mJsepTransceiver->mRecvTrack.GetStreamIds();
+
+  if (mRemoteSetSendBit != mJsepTransceiver->mRecvTrack.GetRemoteSetSendBit()) {
+    mRemoteSetSendBit = mJsepTransceiver->mRecvTrack.GetRemoteSetSendBit();
+    if (mRemoteSetSendBit) {
+      needsTrackEvent = true;
+    } else {
+      aChanges->mTracksToMute.push_back(mTrack);
+    }
+  }
+
+  if (needsTrackEvent) {
+    aChanges->mTrackEvents.push_back({this, mStreamIds});
+  }
+}
+
+void RTCRtpReceiver::MozAddRIDExtension(unsigned short aExtensionId) {
+  if (mPipeline) {
+    mPipeline->AddRIDExtension_m(aExtensionId);
+  }
+}
+
+void RTCRtpReceiver::MozAddRIDFilter(const nsAString& aRid) {
+  if (mPipeline) {
+    mPipeline->AddRIDFilter_m(NS_ConvertUTF16toUTF8(aRid).get());
+  }
+}
+
+// test-only: adds fake CSRCs and audio data
+void RTCRtpReceiver::MozInsertAudioLevelForContributingSource(
+    const uint32_t aSource, const DOMHighResTimeStamp aTimestamp,
+    const uint32_t aRtpTimestamp, const bool aHasLevel, const uint8_t aLevel) {
+  if (!mPipeline || mPipeline->IsVideo() || !mPipeline->Conduit()) {
+    return;
+  }
+  WebrtcAudioConduit* audio_conduit =
+      static_cast<WebrtcAudioConduit*>(mPipeline->Conduit());
+  audio_conduit->InsertAudioLevelForContributingSource(
+      aSource, aTimestamp, aRtpTimestamp, aHasLevel, aLevel);
+}
+
+void RTCRtpReceiver::OnRtcpBye() { SetReceiveTrackMuted(true); }
+
+void RTCRtpReceiver::OnRtcpTimeout() { SetReceiveTrackMuted(true); }
+
+void RTCRtpReceiver::SetReceiveTrackMuted(bool aMuted) {
+  if (mTrack) {
+    // This sets the muted state for mTrack and all its clones.
+    static_cast<RemoteTrackSource&>(mTrack->GetSource()).SetMuted(aMuted);
+  }
+}
+
+std::string RTCRtpReceiver::GetMid() const {
+  if (mJsepTransceiver->IsAssociated()) {
+    return mJsepTransceiver->GetMid();
+  }
+  return std::string();
+}
+
+}  // namespace mozilla::dom
+
+#undef LOGTAG