Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 1046 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..efe83fb782
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
@@ -0,0 +1,1046 @@
+/* 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 <stdint.h>
+
+#include <vector>
+#include <string>
+#include <set>
+
+#include "call/call.h"
+#include "call/audio_receive_stream.h"
+#include "call/video_receive_stream.h"
+#include "api/rtp_parameters.h"
+#include "api/units/timestamp.h"
+#include "api/units/time_delta.h"
+#include "system_wrappers/include/clock.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+#include "RTCRtpTransceiver.h"
+#include "PeerConnectionImpl.h"
+#include "RTCStatsReport.h"
+#include "mozilla/dom/RTCRtpReceiverBinding.h"
+#include "mozilla/dom/RTCRtpSourcesBinding.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "jsep/JsepTransceiver.h"
+#include "libwebrtcglue/MediaConduitControl.h"
+#include "libwebrtcglue/MediaConduitInterface.h"
+#include "transportbridge/MediaPipeline.h"
+#include "sdp/SdpEnum.h"
+#include "sdp/SdpAttribute.h"
+#include "MediaTransportHandler.h"
+#include "RemoteTrackSource.h"
+
+#include "mozilla/dom/RTCRtpCapabilitiesBinding.h"
+#include "mozilla/dom/MediaStreamTrack.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/Nullable.h"
+#include "mozilla/dom/Document.h"
+#include "mozilla/dom/AudioStreamTrack.h"
+#include "mozilla/dom/VideoStreamTrack.h"
+#include "mozilla/dom/RTCRtpScriptTransform.h"
+
+#include "nsPIDOMWindow.h"
+#include "PrincipalHandle.h"
+#include "nsIPrincipal.h"
+#include "MediaTrackGraph.h"
+#include "nsStringFwd.h"
+#include "MediaSegment.h"
+#include "nsLiteralString.h"
+#include "nsTArray.h"
+#include "nsDOMNavigationTiming.h"
+#include "MainThreadUtils.h"
+#include "ErrorList.h"
+#include "nsWrapperCache.h"
+#include "nsISupports.h"
+#include "nsCOMPtr.h"
+#include "nsIScriptObjectPrincipal.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsDebug.h"
+#include "nsThreadUtils.h"
+#include "PerformanceRecorder.h"
+
+#include "mozilla/NullPrincipal.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/Logging.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/AbstractThread.h"
+#include "mozilla/StateWatching.h"
+#include "mozilla/Maybe.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/AlreadyAddRefed.h"
+#include "mozilla/MozPromise.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/fallible.h"
+#include "mozilla/mozalloc_oom.h"
+#include "mozilla/ErrorResult.h"
+#include "js/RootingAPI.h"
+
+namespace mozilla::dom {
+
+LazyLogModule gReceiverLog("RTCRtpReceiver");
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpReceiver)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpReceiver)
+  tmp->Unlink();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(RTCRtpReceiver)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mWindow, mPc, mTransceiver, mTransform,
+                                    mTrack, mTrackSource)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+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
+
+static PrincipalHandle GetPrincipalHandle(nsPIDOMWindowInner* aWindow,
+                                          PrincipalPrivacy aPrivacy) {
+  // 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 (aPrivacy == PrincipalPrivacy::Private) {
+    principal = NullPrincipal::CreateWithInheritedAttributes(principal);
+  }
+  return MakePrincipalHandle(principal);
+}
+
+#define INIT_CANONICAL(name, val)         \
+  name(AbstractThread::MainThread(), val, \
+       "RTCRtpReceiver::" #name " (Canonical)")
+
+RTCRtpReceiver::RTCRtpReceiver(
+    nsPIDOMWindowInner* aWindow, PrincipalPrivacy aPrivacy,
+    PeerConnectionImpl* aPc, MediaTransportHandler* aTransportHandler,
+    AbstractThread* aCallThread, nsISerialEventTarget* aStsThread,
+    MediaSessionConduit* aConduit, RTCRtpTransceiver* aTransceiver,
+    const TrackingId& aTrackingId)
+    : mWatchManager(this, AbstractThread::MainThread()),
+      mWindow(aWindow),
+      mPc(aPc),
+      mCallThread(aCallThread),
+      mStsThread(aStsThread),
+      mTransportHandler(aTransportHandler),
+      mTransceiver(aTransceiver),
+      INIT_CANONICAL(mSsrc, 0),
+      INIT_CANONICAL(mVideoRtxSsrc, 0),
+      INIT_CANONICAL(mLocalRtpExtensions, RtpExtList()),
+      INIT_CANONICAL(mAudioCodecs, std::vector<AudioCodecConfig>()),
+      INIT_CANONICAL(mVideoCodecs, std::vector<VideoCodecConfig>()),
+      INIT_CANONICAL(mVideoRtpRtcpConfig, Nothing()),
+      INIT_CANONICAL(mReceiving, false),
+      INIT_CANONICAL(mFrameTransformerProxy, nullptr) {
+  PrincipalHandle principalHandle = GetPrincipalHandle(aWindow, aPrivacy);
+  const bool isAudio = aConduit->type() == MediaSessionConduit::AUDIO;
+
+  MediaTrackGraph* graph = MediaTrackGraph::GetInstance(
+      isAudio ? MediaTrackGraph::AUDIO_THREAD_DRIVER
+              : MediaTrackGraph::SYSTEM_THREAD_DRIVER,
+      aWindow, MediaTrackGraph::REQUEST_DEFAULT_SAMPLE_RATE,
+      MediaTrackGraph::DEFAULT_OUTPUT_DEVICE);
+
+  if (isAudio) {
+    auto* source = graph->CreateSourceTrack(MediaSegment::AUDIO);
+    mTrackSource = MakeAndAddRef<RemoteTrackSource>(
+        source, this, principalHandle, u"remote audio"_ns, aTrackingId);
+    mTrack = MakeAndAddRef<AudioStreamTrack>(aWindow, source, mTrackSource);
+    mPipeline = MakeAndAddRef<MediaPipelineReceiveAudio>(
+        mPc->GetHandle(), aTransportHandler, aCallThread, mStsThread.get(),
+        *aConduit->AsAudioSessionConduit(), mTrackSource->Stream(), aTrackingId,
+        principalHandle, aPrivacy);
+  } else {
+    auto* source = graph->CreateSourceTrack(MediaSegment::VIDEO);
+    mTrackSource = MakeAndAddRef<RemoteTrackSource>(
+        source, this, principalHandle, u"remote video"_ns, aTrackingId);
+    mTrack = MakeAndAddRef<VideoStreamTrack>(aWindow, source, mTrackSource);
+    mPipeline = MakeAndAddRef<MediaPipelineReceiveVideo>(
+        mPc->GetHandle(), aTransportHandler, aCallThread, mStsThread.get(),
+        *aConduit->AsVideoSessionConduit(), mTrackSource->Stream(), aTrackingId,
+        principalHandle, aPrivacy);
+  }
+
+  mPipeline->InitControl(this);
+
+  // Spec says remote tracks start out muted.
+  mTrackSource->SetMuted(true);
+
+  // 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)) {
+    mRtcpByeListener = aConduit->RtcpByeEvent().Connect(
+        GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtcpBye);
+    mRtcpTimeoutListener = aConduit->RtcpTimeoutEvent().Connect(
+        GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtcpTimeout);
+  }
+
+  mWatchManager.Watch(mReceiveTrackMute,
+                      &RTCRtpReceiver::UpdateReceiveTrackMute);
+}
+
+#undef INIT_CANONICAL
+
+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 (!mTransceiver) {
+    return nullptr;
+  }
+  return mTransceiver->GetDtlsTransport();
+}
+
+void RTCRtpReceiver::GetCapabilities(
+    const GlobalObject&, const nsAString& aKind,
+    Nullable<dom::RTCRtpCapabilities>& aResult) {
+  PeerConnectionImpl::GetCapabilities(aKind, aResult, sdp::Direction::kRecv);
+}
+
+already_AddRefed<Promise> RTCRtpReceiver::GetStats(ErrorResult& aError) {
+  nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(mWindow);
+  RefPtr<Promise> promise = Promise::Create(global, aError);
+  if (NS_WARN_IF(aError.Failed())) {
+    return nullptr;
+  }
+
+  if (NS_WARN_IF(!mTransceiver)) {
+    // TODO(bug 1056433): When we stop nulling this out when the PC is closed
+    // (or when the transceiver is stopped), we can remove this code. We
+    // resolve instead of reject in order to make this eventual change in
+    // behavior a little smaller.
+    promise->MaybeResolve(new RTCStatsReport(mWindow));
+    return promise.forget();
+  }
+
+  mTransceiver->ChainToDomPromiseWithCodecStats(GetStatsInternal(), promise);
+  return promise.forget();
+}
+
+nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal(
+    bool aSkipIceStats) {
+  MOZ_ASSERT(NS_IsMainThread());
+  nsTArray<RefPtr<RTCStatsPromise>> promises(3);
+
+  if (!mPipeline) {
+    return promises;
+  }
+
+  if (!mHaveStartedReceiving) {
+    return promises;
+  }
+
+  nsString recvTrackId;
+  MOZ_ASSERT(mTrack);
+  if (mTrack) {
+    mTrack->GetId(recvTrackId);
+  }
+
+  {
+    // Add bandwidth estimation stats
+    promises.AppendElement(InvokeAsync(
+        mCallThread, __func__,
+        [conduit = mPipeline->mConduit, recvTrackId]() mutable {
+          auto report = MakeUnique<dom::RTCStatsCollection>();
+          const Maybe<webrtc::Call::Stats> stats = conduit->GetCallStats();
+          stats.apply([&](const auto& aStats) {
+            dom::RTCBandwidthEstimationInternal bw;
+            bw.mTrackIdentifier = recvTrackId;
+            bw.mSendBandwidthBps.Construct(aStats.send_bandwidth_bps / 8);
+            bw.mMaxPaddingBps.Construct(aStats.max_padding_bitrate_bps / 8);
+            bw.mReceiveBandwidthBps.Construct(aStats.recv_bandwidth_bps / 8);
+            bw.mPacerDelayMs.Construct(aStats.pacer_delay_ms);
+            if (aStats.rtt_ms >= 0) {
+              bw.mRttMs.Construct(aStats.rtt_ms);
+            }
+            if (!report->mBandwidthEstimations.AppendElement(std::move(bw),
+                                                             fallible)) {
+              mozalloc_handle_oom(0);
+            }
+          });
+          return RTCStatsPromise::CreateAndResolve(std::move(report), __func__);
+        }));
+  }
+
+  promises.AppendElement(
+      InvokeAsync(
+          mCallThread, __func__,
+          [pipeline = mPipeline, recvTrackId] {
+            auto report = MakeUnique<dom::RTCStatsCollection>();
+            auto asAudio = pipeline->mConduit->AsAudioSessionConduit();
+            auto asVideo = pipeline->mConduit->AsVideoSessionConduit();
+
+            nsString kind = asVideo.isNothing() ? u"audio"_ns : u"video"_ns;
+            nsString idstr = kind + u"_"_ns;
+            idstr.AppendInt(static_cast<uint32_t>(pipeline->Level()));
+
+            Maybe<uint32_t> ssrc = pipeline->mConduit->GetRemoteSSRC();
+
+            // Add frame history
+            asVideo.apply([&](const auto& conduit) {
+              if (conduit->AddFrameHistory(&report->mVideoFrameHistories)) {
+                auto& history = report->mVideoFrameHistories.LastElement();
+                history.mTrackIdentifier = recvTrackId;
+              }
+            });
+
+            // 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;
+
+            auto constructCommonRemoteOutboundRtpStats =
+                [&](RTCRemoteOutboundRtpStreamStats& aRemote,
+                    const DOMHighResTimeStamp& aTimestamp) {
+                  remoteId = u"inbound_rtcp_"_ns + idstr;
+                  aRemote.mTimestamp.Construct(aTimestamp);
+                  aRemote.mId.Construct(remoteId);
+                  aRemote.mType.Construct(RTCStatsType::Remote_outbound_rtp);
+                  ssrc.apply([&](uint32_t aSsrc) { aRemote.mSsrc = aSsrc; });
+                  aRemote.mKind = kind;
+                  aRemote.mMediaType.Construct(
+                      kind);  // mediaType is the old name for kind.
+                  aRemote.mLocalId.Construct(localId);
+                };
+
+            auto constructCommonInboundRtpStats =
+                [&](RTCInboundRtpStreamStats& aLocal) {
+                  aLocal.mTrackIdentifier = recvTrackId;
+                  aLocal.mTimestamp.Construct(
+                      pipeline->GetTimestampMaker().GetNow().ToDom());
+                  aLocal.mId.Construct(localId);
+                  aLocal.mType.Construct(RTCStatsType::Inbound_rtp);
+                  ssrc.apply([&](uint32_t aSsrc) { aLocal.mSsrc = aSsrc; });
+                  aLocal.mKind = kind;
+                  aLocal.mMediaType.Construct(
+                      kind);  // mediaType is the old name for kind.
+                  if (remoteId.Length()) {
+                    aLocal.mRemoteId.Construct(remoteId);
+                  }
+                };
+
+            asAudio.apply([&](auto& aConduit) {
+              Maybe<webrtc::AudioReceiveStreamInterface::Stats> audioStats =
+                  aConduit->GetReceiverStats();
+              if (audioStats.isNothing()) {
+                return;
+              }
+
+              if (!audioStats->last_packet_received.has_value()) {
+                // By spec: "The lifetime of all RTP monitored objects starts
+                // when the RTP stream is first used: When the first RTP packet
+                // is sent or received on the SSRC it represents"
+                return;
+              }
+
+              // First, fill in remote stat with rtcp sender data, if present.
+              if (audioStats->last_sender_report_timestamp_ms) {
+                RTCRemoteOutboundRtpStreamStats remote;
+                constructCommonRemoteOutboundRtpStats(
+                    remote,
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            *audioStats->last_sender_report_timestamp_ms) +
+                            webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ToDom());
+                remote.mPacketsSent.Construct(
+                    audioStats->sender_reports_packets_sent);
+                remote.mBytesSent.Construct(
+                    audioStats->sender_reports_bytes_sent);
+                remote.mRemoteTimestamp.Construct(
+                    *audioStats->last_sender_report_remote_timestamp_ms);
+                if (!report->mRemoteOutboundRtpStreamStats.AppendElement(
+                        std::move(remote), fallible)) {
+                  mozalloc_handle_oom(0);
+                }
+              }
+
+              // Then, fill in local side (with cross-link to remote only if
+              // present)
+              RTCInboundRtpStreamStats local;
+              constructCommonInboundRtpStats(local);
+              local.mJitter.Construct(audioStats->jitter_ms / 1000.0);
+              local.mPacketsLost.Construct(audioStats->packets_lost);
+              local.mPacketsReceived.Construct(audioStats->packets_received);
+              local.mPacketsDiscarded.Construct(audioStats->packets_discarded);
+              local.mBytesReceived.Construct(
+                  audioStats->payload_bytes_received);
+              // Always missing from libwebrtc stats
+              // if (audioStats->estimated_playout_ntp_timestamp_ms) {
+              //   local.mEstimatedPlayoutTimestamp.Construct(
+              //       RTCStatsTimestamp::FromNtp(
+              //           aConduit->GetTimestampMaker(),
+              //           webrtc::Timestamp::Millis(
+              //               *audioStats->estimated_playout_ntp_timestamp_ms))
+              //           .ToDom());
+              // }
+              local.mJitterBufferDelay.Construct(
+                  audioStats->jitter_buffer_delay_seconds);
+              local.mJitterBufferEmittedCount.Construct(
+                  audioStats->jitter_buffer_emitted_count);
+              local.mTotalSamplesReceived.Construct(
+                  audioStats->total_samples_received);
+              local.mConcealedSamples.Construct(audioStats->concealed_samples);
+              local.mSilentConcealedSamples.Construct(
+                  audioStats->silent_concealed_samples);
+              if (audioStats->last_packet_received.has_value()) {
+                local.mLastPacketReceivedTimestamp.Construct(
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            audioStats->last_packet_received->ms()) +
+                            webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ToDom());
+              }
+              local.mHeaderBytesReceived.Construct(
+                  audioStats->header_and_padding_bytes_received);
+              local.mFecPacketsReceived.Construct(
+                  audioStats->fec_packets_received);
+              local.mFecPacketsDiscarded.Construct(
+                  audioStats->fec_packets_discarded);
+              local.mConcealmentEvents.Construct(
+                  audioStats->concealment_events);
+
+              local.mInsertedSamplesForDeceleration.Construct(
+                  audioStats->inserted_samples_for_deceleration);
+              local.mRemovedSamplesForAcceleration.Construct(
+                  audioStats->removed_samples_for_acceleration);
+              if (audioStats->audio_level >= 0 &&
+                  audioStats->audio_level <= 32767) {
+                local.mAudioLevel.Construct(audioStats->audio_level / 32767.0);
+              }
+              local.mTotalAudioEnergy.Construct(
+                  audioStats->total_output_energy);
+              local.mTotalSamplesDuration.Construct(
+                  audioStats->total_output_duration);
+
+              if (!report->mInboundRtpStreamStats.AppendElement(
+                      std::move(local), fallible)) {
+                mozalloc_handle_oom(0);
+              }
+            });
+
+            asVideo.apply([&](auto& aConduit) {
+              Maybe<webrtc::VideoReceiveStreamInterface::Stats> videoStats =
+                  aConduit->GetReceiverStats();
+              if (videoStats.isNothing()) {
+                return;
+              }
+
+              if (!videoStats->rtp_stats.last_packet_received.has_value()) {
+                // By spec: "The lifetime of all RTP monitored objects starts
+                // when the RTP stream is first used: When the first RTP packet
+                // is sent or received on the SSRC it represents"
+                return;
+              }
+
+              // First, fill in remote stat with rtcp sender data, if present.
+              if (videoStats->rtcp_sender_ntp_timestamp_ms) {
+                RTCRemoteOutboundRtpStreamStats remote;
+                constructCommonRemoteOutboundRtpStats(
+                    remote, RTCStatsTimestamp::FromNtp(
+                                aConduit->GetTimestampMaker(),
+                                webrtc::Timestamp::Millis(
+                                    videoStats->rtcp_sender_ntp_timestamp_ms))
+                                .ToDom());
+                remote.mPacketsSent.Construct(
+                    videoStats->rtcp_sender_packets_sent);
+                remote.mBytesSent.Construct(
+                    videoStats->rtcp_sender_octets_sent);
+                remote.mRemoteTimestamp.Construct(
+                    (webrtc::TimeDelta::Millis(
+                         videoStats->rtcp_sender_remote_ntp_timestamp_ms) -
+                     webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ms());
+                if (!report->mRemoteOutboundRtpStreamStats.AppendElement(
+                        std::move(remote), fallible)) {
+                  mozalloc_handle_oom(0);
+                }
+              }
+
+              // Then, fill in local side (with cross-link to remote only if
+              // present)
+              RTCInboundRtpStreamStats local;
+              constructCommonInboundRtpStats(local);
+              local.mJitter.Construct(
+                  static_cast<double>(videoStats->rtp_stats.jitter) /
+                  webrtc::kVideoPayloadTypeFrequency);
+              local.mPacketsLost.Construct(videoStats->rtp_stats.packets_lost);
+              local.mPacketsReceived.Construct(
+                  videoStats->rtp_stats.packet_counter.packets);
+              local.mPacketsDiscarded.Construct(videoStats->packets_discarded);
+              local.mDiscardedPackets.Construct(videoStats->packets_discarded);
+              local.mBytesReceived.Construct(
+                  videoStats->rtp_stats.packet_counter.payload_bytes);
+
+              // Fill in packet type statistics
+              local.mNackCount.Construct(
+                  videoStats->rtcp_packet_type_counts.nack_packets);
+              local.mFirCount.Construct(
+                  videoStats->rtcp_packet_type_counts.fir_packets);
+              local.mPliCount.Construct(
+                  videoStats->rtcp_packet_type_counts.pli_packets);
+
+              // Lastly, fill in video decoder stats
+              local.mFramesDecoded.Construct(videoStats->frames_decoded);
+
+              local.mFramesPerSecond.Construct(videoStats->decode_frame_rate);
+              local.mFrameWidth.Construct(videoStats->width);
+              local.mFrameHeight.Construct(videoStats->height);
+              // XXX: key_frames + delta_frames may undercount frames because
+              // they were dropped in FrameBuffer::InsertFrame. (bug 1766553)
+              local.mFramesReceived.Construct(
+                  videoStats->frame_counts.key_frames +
+                  videoStats->frame_counts.delta_frames);
+              local.mJitterBufferDelay.Construct(
+                  videoStats->jitter_buffer_delay.seconds<double>());
+              local.mJitterBufferEmittedCount.Construct(
+                  videoStats->jitter_buffer_emitted_count);
+
+              if (videoStats->qp_sum) {
+                local.mQpSum.Construct(videoStats->qp_sum.value());
+              }
+              local.mTotalDecodeTime.Construct(
+                  double(videoStats->total_decode_time.ms()) / 1000);
+              local.mTotalInterFrameDelay.Construct(
+                  videoStats->total_inter_frame_delay);
+              local.mTotalSquaredInterFrameDelay.Construct(
+                  videoStats->total_squared_inter_frame_delay);
+              if (videoStats->rtp_stats.last_packet_received.has_value()) {
+                local.mLastPacketReceivedTimestamp.Construct(
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            videoStats->rtp_stats.last_packet_received->ms()) +
+                            webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ToDom());
+              }
+              local.mHeaderBytesReceived.Construct(
+                  videoStats->rtp_stats.packet_counter.header_bytes +
+                  videoStats->rtp_stats.packet_counter.padding_bytes);
+              local.mTotalProcessingDelay.Construct(
+                  videoStats->total_processing_delay.seconds<double>());
+              /*
+               * Potential new stats that are now available upstream
+                   .if (videoStats->estimated_playout_ntp_timestamp_ms) {
+                local.mEstimatedPlayoutTimestamp.Construct(
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            *videoStats->estimated_playout_ntp_timestamp_ms))
+                        .ToDom());
+              }
+               */
+              // Not including frames dropped in the rendering pipe, which
+              // is not of webrtc's concern anyway?!
+              local.mFramesDropped.Construct(videoStats->frames_dropped);
+              if (!report->mInboundRtpStreamStats.AppendElement(
+                      std::move(local), fallible)) {
+                mozalloc_handle_oom(0);
+              }
+            });
+            return RTCStatsPromise::CreateAndResolve(std::move(report),
+                                                     __func__);
+          })
+          ->Then(
+              mStsThread, __func__,
+              [pipeline = mPipeline](UniquePtr<RTCStatsCollection> aReport) {
+                // Fill in Contributing Source statistics
+                if (!aReport->mInboundRtpStreamStats.IsEmpty() &&
+                    aReport->mInboundRtpStreamStats[0].mId.WasPassed()) {
+                  pipeline->GetContributingSourceStats(
+                      aReport->mInboundRtpStreamStats[0].mId.Value(),
+                      aReport->mRtpContributingSourceStats);
+                }
+                return RTCStatsPromise::CreateAndResolve(std::move(aReport),
+                                                         __func__);
+              },
+              [] {
+                MOZ_CRASH("Unexpected reject");
+                return RTCStatsPromise::CreateAndReject(NS_ERROR_UNEXPECTED,
+                                                        __func__);
+              }));
+
+  if (!aSkipIceStats && GetJsepTransceiver().mTransport.mComponents) {
+    promises.AppendElement(mTransportHandler->GetIceStats(
+        GetJsepTransceiver().mTransport.mTransportId,
+        mPipeline->GetTimestampMaker().GetNow().ToDom()));
+  }
+
+  return promises;
+}
+
+void RTCRtpReceiver::SetJitterBufferTarget(
+    const Nullable<DOMHighResTimeStamp>& aTargetMs, ErrorResult& aError) {
+  // Spec says jitter buffer target cannot be negative or larger than 4000
+  // milliseconds and to throw RangeError if it is. If an invalid value is
+  // received we return early to preserve the current JitterBufferTarget
+  // internal slot and jitter buffer values.
+  if (mPipeline && mPipeline->mConduit) {
+    if (!aTargetMs.IsNull() &&
+        (aTargetMs.Value() < 0.0 || aTargetMs.Value() > 4000.0)) {
+      aError.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("jitterBufferTarget");
+      return;
+    }
+
+    mJitterBufferTarget.reset();
+
+    if (!aTargetMs.IsNull()) {
+      mJitterBufferTarget = Some(aTargetMs.Value());
+    }
+    // If aJitterBufferTarget is null then we are resetting the jitter buffer so
+    // pass the default target of 0.0.
+    mPipeline->mConduit->SetJitterBufferTarget(
+        mJitterBufferTarget.valueOr(0.0));
+  }
+}
+
+void RTCRtpReceiver::GetContributingSources(
+    nsTArray<RTCRtpContributingSource>& aSources) {
+  // Duplicate code...
+  if (mPipeline && mPipeline->mConduit) {
+    nsTArray<dom::RTCRtpSourceEntry> sources;
+    mPipeline->mConduit->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->mConduit) {
+    nsTArray<dom::RTCRtpSourceEntry> sources;
+    mPipeline->mConduit->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() {
+  MOZ_ASSERT(NS_IsMainThread());
+  mWatchManager.Shutdown();
+  if (mPipeline) {
+    mPipeline->Shutdown();
+    mPipeline = nullptr;
+  }
+  if (mTrackSource) {
+    mTrackSource->Destroy();
+  }
+  mCallThread = nullptr;
+  mRtcpByeListener.DisconnectIfExists();
+  mRtcpTimeoutListener.DisconnectIfExists();
+  mUnmuteListener.DisconnectIfExists();
+  if (mTransform) {
+    mTransform->GetProxy().SetReceiver(nullptr);
+  }
+}
+
+void RTCRtpReceiver::BreakCycles() {
+  mWindow = nullptr;
+  mPc = nullptr;
+  mTrack = nullptr;
+  mTrackSource = nullptr;
+}
+
+void RTCRtpReceiver::Unlink() {
+  if (mTransceiver) {
+    mTransceiver->Unlink();
+  }
+}
+
+void RTCRtpReceiver::UpdateTransport() {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (!mHaveSetupTransport) {
+    mPipeline->SetLevel(GetJsepTransceiver().GetLevel());
+    mHaveSetupTransport = true;
+  }
+
+  UniquePtr<MediaPipelineFilter> filter;
+
+  auto const& details = GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails();
+  if (GetJsepTransceiver().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 : GetJsepTransceiver().mRecvTrack.GetSsrcs()) {
+      filter->AddRemoteSSRC(ssrc);
+    }
+    for (uint32_t ssrc : GetJsepTransceiver().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 = GetJsepTransceiver()
+                         .mRecvTrack.GetNegotiatedDetails()
+                         ->GetUniquePayloadTypes();
+    for (unsigned char& uniquePt : uniquePts) {
+      filter->AddUniquePT(uniquePt);
+    }
+  }
+
+  mPipeline->UpdateTransport_m(GetJsepTransceiver().mTransport.mTransportId,
+                               std::move(filter));
+}
+
+void RTCRtpReceiver::UpdateConduit() {
+  if (mPipeline->mConduit->type() == MediaSessionConduit::VIDEO) {
+    UpdateVideoConduit();
+  } else {
+    UpdateAudioConduit();
+  }
+
+  if ((mReceiving = mTransceiver->IsReceiving())) {
+    mHaveStartedReceiving = true;
+  }
+}
+
+void RTCRtpReceiver::UpdateVideoConduit() {
+  RefPtr<VideoSessionConduit> conduit =
+      *mPipeline->mConduit->AsVideoSessionConduit();
+
+  // 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 (!GetJsepTransceiver().mRecvTrack.GetSsrcs().empty()) {
+    MOZ_LOG(gReceiverLog, LogLevel::Debug,
+            ("%s[%s]: %s Setting remote SSRC %u", mPc->GetHandle().c_str(),
+             GetMid().c_str(), __FUNCTION__,
+             GetJsepTransceiver().mRecvTrack.GetSsrcs().front()));
+    uint32_t rtxSsrc =
+        GetJsepTransceiver().mRecvTrack.GetRtxSsrcs().empty()
+            ? 0
+            : GetJsepTransceiver().mRecvTrack.GetRtxSsrcs().front();
+    mSsrc = GetJsepTransceiver().mRecvTrack.GetSsrcs().front();
+    mVideoRtxSsrc = 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.
+    // In any case, do not disable SSRC changes if no SSRCs were negotiated
+    if (GetJsepTransceiver().HasBundleLevel() &&
+        (!GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() ||
+         !GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails()->GetExt(
+             webrtc::RtpExtension::kMidUri))) {
+      mCallThread->Dispatch(
+          NewRunnableMethod("VideoSessionConduit::DisableSsrcChanges", conduit,
+                            &VideoSessionConduit::DisableSsrcChanges));
+    }
+  }
+
+  if (GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() &&
+      GetJsepTransceiver().mRecvTrack.GetActive()) {
+    const auto& details(
+        *GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails());
+
+    {
+      std::vector<webrtc::RtpExtension> extmaps;
+      // @@NG read extmap from track
+      details.ForEachRTPHeaderExtension(
+          [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
+            extmaps.emplace_back(extmap.extensionname, extmap.entry);
+          });
+      mLocalRtpExtensions = extmaps;
+    }
+
+    std::vector<VideoCodecConfig> configs;
+    RTCRtpTransceiver::NegotiatedDetailsToVideoCodecConfigs(details, &configs);
+    if (configs.empty()) {
+      // TODO: Are we supposed to plumb this error back to JS? This does not
+      // seem like a failure to set an answer, it just means that codec
+      // negotiation failed. For now, we're just doing the same thing we do
+      // if negotiation as a whole failed.
+      MOZ_LOG(gReceiverLog, LogLevel::Error,
+              ("%s[%s]: %s  No video codecs were negotiated (recv).",
+               mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
+      return;
+    }
+
+    mVideoCodecs = configs;
+    mVideoRtpRtcpConfig = Some(details.GetRtpRtcpConfig());
+  }
+}
+
+void RTCRtpReceiver::UpdateAudioConduit() {
+  RefPtr<AudioSessionConduit> conduit =
+      *mPipeline->mConduit->AsAudioSessionConduit();
+
+  if (!GetJsepTransceiver().mRecvTrack.GetSsrcs().empty()) {
+    MOZ_LOG(gReceiverLog, LogLevel::Debug,
+            ("%s[%s]: %s Setting remote SSRC %u", mPc->GetHandle().c_str(),
+             GetMid().c_str(), __FUNCTION__,
+             GetJsepTransceiver().mRecvTrack.GetSsrcs().front()));
+    mSsrc = GetJsepTransceiver().mRecvTrack.GetSsrcs().front();
+
+    // 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.
+    // In any case, do not disable SSRC changes if no SSRCs were negotiated
+    if (GetJsepTransceiver().HasBundleLevel() &&
+        (!GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() ||
+         !GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails()->GetExt(
+             webrtc::RtpExtension::kMidUri))) {
+      mCallThread->Dispatch(
+          NewRunnableMethod("AudioSessionConduit::DisableSsrcChanges", conduit,
+                            &AudioSessionConduit::DisableSsrcChanges));
+    }
+  }
+
+  if (GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() &&
+      GetJsepTransceiver().mRecvTrack.GetActive()) {
+    const auto& details(
+        *GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails());
+    std::vector<AudioCodecConfig> configs;
+    RTCRtpTransceiver::NegotiatedDetailsToAudioCodecConfigs(details, &configs);
+    if (configs.empty()) {
+      // TODO: Are we supposed to plumb this error back to JS? This does not
+      // seem like a failure to set an answer, it just means that codec
+      // negotiation failed. For now, we're just doing the same thing we do
+      // if negotiation as a whole failed.
+      MOZ_LOG(gReceiverLog, LogLevel::Error,
+              ("%s[%s]: %s No audio codecs were negotiated (recv)",
+               mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
+      return;
+    }
+
+    // Ensure conduit knows about extensions prior to creating streams
+    {
+      std::vector<webrtc::RtpExtension> extmaps;
+      // @@NG read extmap from track
+      details.ForEachRTPHeaderExtension(
+          [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
+            extmaps.emplace_back(extmap.extensionname, extmap.entry);
+          });
+      mLocalRtpExtensions = extmaps;
+    }
+
+    mAudioCodecs = configs;
+  }
+}
+
+void RTCRtpReceiver::Stop() {
+  MOZ_ASSERT(mTransceiver->Stopped() || mTransceiver->Stopping());
+  mReceiving = false;
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__, [trackSource = mTrackSource] { trackSource->ForceEnded(); }));
+}
+
+bool RTCRtpReceiver::HasTrack(const dom::MediaStreamTrack* aTrack) const {
+  return !aTrack || (mTrack == aTrack);
+}
+
+void RTCRtpReceiver::SyncFromJsep(const JsepTransceiver& aJsepTransceiver) {
+  if (!mPipeline) {
+    return;
+  }
+
+  // Spec says we set [[Receptive]] to true on sLD(sendrecv/recvonly), and to
+  // false on sRD(recvonly/inactive), sLD(sendonly/inactive), or when stop()
+  // is called.
+  bool wasReceptive = mReceptive;
+  mReceptive = aJsepTransceiver.mRecvTrack.GetReceptive();
+  if (!wasReceptive && mReceptive) {
+    mUnmuteListener = mPipeline->mConduit->RtpPacketEvent().Connect(
+        GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtpPacket);
+  } else if (wasReceptive && !mReceptive) {
+    mUnmuteListener.DisconnectIfExists();
+  }
+}
+
+void RTCRtpReceiver::SyncToJsep(JsepTransceiver& aJsepTransceiver) const {}
+
+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(
+      GetJsepTransceiver().mRecvTrack.GetStreamIds().begin(),
+      GetJsepTransceiver().mRecvTrack.GetStreamIds().end());
+  MOZ_ASSERT(GetJsepTransceiver().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 : GetJsepTransceiver().mRecvTrack.GetStreamIds()) {
+    if (!oldIds.count(id)) {
+      needsTrackEvent = true;
+      aChanges->mStreamAssociationsAdded.push_back({mTrack, id});
+    }
+  }
+
+  mStreamIds = GetJsepTransceiver().mRecvTrack.GetStreamIds();
+
+  if (mRemoteSetSendBit !=
+      GetJsepTransceiver().mRecvTrack.GetRemoteSetSendBit()) {
+    mRemoteSetSendBit = GetJsepTransceiver().mRecvTrack.GetRemoteSetSendBit();
+    if (mRemoteSetSendBit) {
+      needsTrackEvent = true;
+    } else {
+      aChanges->mReceiversToMute.push_back(this);
+    }
+  }
+
+  if (needsTrackEvent) {
+    aChanges->mTrackEvents.push_back({this, mStreamIds});
+  }
+}
+
+void RTCRtpReceiver::UpdatePrincipalPrivacy(PrincipalPrivacy aPrivacy) {
+  if (!mPipeline) {
+    return;
+  }
+
+  if (aPrivacy != PrincipalPrivacy::Private) {
+    return;
+  }
+
+  mPipeline->SetPrivatePrincipal(GetPrincipalHandle(mWindow, aPrivacy));
+}
+
+// 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->mConduit) {
+    return;
+  }
+  mPipeline->mConduit->InsertAudioLevelForContributingSource(
+      aSource, aTimestamp, aRtpTimestamp, aHasLevel, aLevel);
+}
+
+void RTCRtpReceiver::OnRtcpBye() { mReceiveTrackMute = true; }
+
+void RTCRtpReceiver::OnRtcpTimeout() { mReceiveTrackMute = true; }
+
+void RTCRtpReceiver::SetTrackMuteFromRemoteSdp() {
+  MOZ_ASSERT(!mReceptive,
+             "PeerConnectionImpl should have blocked unmute events prior to "
+             "firing mute");
+  mReceiveTrackMute = true;
+  // Set the mute state (and fire the mute event) synchronously. Unmute is
+  // handled asynchronously after receiving RTP packets.
+  UpdateReceiveTrackMute();
+  MOZ_ASSERT(mTrack->Muted(), "Muted state was indeed set synchronously");
+}
+
+void RTCRtpReceiver::OnRtpPacket() {
+  MOZ_ASSERT(mReceptive, "We should not be registered unless this is set!");
+  // We should be registered since we're currently getting a callback.
+  mUnmuteListener.Disconnect();
+  if (mReceptive) {
+    mReceiveTrackMute = false;
+  }
+}
+
+void RTCRtpReceiver::UpdateReceiveTrackMute() {
+  if (!mTrack) {
+    return;
+  }
+  if (!mTrackSource) {
+    return;
+  }
+  // This sets the muted state for mTrack and all its clones.
+  // Idempotent -- only reacts to changes.
+  mTrackSource->SetMuted(mReceiveTrackMute);
+}
+
+std::string RTCRtpReceiver::GetMid() const {
+  return mTransceiver->GetMidAscii();
+}
+
+JsepTransceiver& RTCRtpReceiver::GetJsepTransceiver() {
+  MOZ_ASSERT(mTransceiver);
+  return mTransceiver->GetJsepTransceiver();
+}
+
+const JsepTransceiver& RTCRtpReceiver::GetJsepTransceiver() const {
+  MOZ_ASSERT(mTransceiver);
+  return mTransceiver->GetJsepTransceiver();
+}
+
+void RTCRtpReceiver::SetTransform(RTCRtpScriptTransform* aTransform,
+                                  ErrorResult& aError) {
+  if (aTransform == mTransform.get()) {
+    // Ok... smile and nod
+    // TODO: Depending on spec, this might throw
+    // https://github.com/w3c/webrtc-encoded-transform/issues/189
+    return;
+  }
+
+  if (aTransform && aTransform->IsClaimed()) {
+    aError.ThrowInvalidStateError("transform has already been used elsewhere");
+    return;
+  }
+
+  if (aTransform) {
+    mFrameTransformerProxy = &aTransform->GetProxy();
+  } else {
+    mFrameTransformerProxy = nullptr;
+  }
+
+  if (mTransform) {
+    mTransform->GetProxy().SetReceiver(nullptr);
+  }
+
+  mTransform = const_cast<RTCRtpScriptTransform*>(aTransform);
+
+  if (mTransform) {
+    mTransform->GetProxy().SetReceiver(this);
+    mTransform->SetClaimed();
+  }
+}
+
+void RTCRtpReceiver::RequestKeyFrame() {
+  if (!mTransform || !mPipeline) {
+    return;
+  }
+
+  mPipeline->mConduit->AsVideoSessionConduit().apply([&](const auto& conduit) {
+    conduit->RequestKeyFrame(&mTransform->GetProxy());
+  });
+}
+
+}  // namespace mozilla::dom
+
+#undef LOGTAG