Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

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

7 files changed, 2473 insertions, 0 deletions

diff --git a/dom/media/webrtc/transportbridge/MediaPipeline.cpp b/dom/media/webrtc/transportbridge/MediaPipeline.cpp
new file mode 100644
index 0000000000..ec861f69bd
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/MediaPipeline.cpp
@@ -0,0 +1,1655 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* 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/. */
+
+// Original author: ekr@rtfm.com
+
+#include "MediaPipeline.h"
+
+#include <inttypes.h>
+#include <math.h>
+#include <sstream>
+
+#include "AudioSegment.h"
+#include "AudioConverter.h"
+#include "DOMMediaStream.h"
+#include "ImageContainer.h"
+#include "ImageTypes.h"
+#include "MediaEngine.h"
+#include "MediaSegment.h"
+#include "MediaTrackGraphImpl.h"
+#include "MediaTrackListener.h"
+#include "MediaStreamTrack.h"
+#include "RtpLogger.h"
+#include "VideoFrameConverter.h"
+#include "VideoSegment.h"
+#include "VideoStreamTrack.h"
+#include "VideoUtils.h"
+#include "mozilla/Logging.h"
+#include "mozilla/NullPrincipal.h"
+#include "mozilla/PeerIdentity.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/SharedThreadPool.h"
+#include "mozilla/Sprintf.h"
+#include "mozilla/StaticPrefs_media.h"
+#include "mozilla/TaskQueue.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/UniquePtrExtensions.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "mozilla/dom/Document.h"
+#include "mozilla/gfx/Point.h"
+#include "mozilla/gfx/Types.h"
+#include "nsError.h"
+#include "nsThreadUtils.h"
+#include "transport/runnable_utils.h"
+#include "jsapi/MediaTransportHandler.h"
+#include "jsapi/PeerConnectionImpl.h"
+#include "Tracing.h"
+#include "libwebrtcglue/WebrtcImageBuffer.h"
+#include "libwebrtcglue/MediaConduitInterface.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+
+// Max size given stereo is 480*2*2 = 1920 (10ms of 16-bits stereo audio at
+// 48KHz)
+#define AUDIO_SAMPLE_BUFFER_MAX_BYTES (480 * 2 * 2)
+static_assert((WEBRTC_MAX_SAMPLE_RATE / 100) * sizeof(uint16_t) * 2 <=
+                  AUDIO_SAMPLE_BUFFER_MAX_BYTES,
+              "AUDIO_SAMPLE_BUFFER_MAX_BYTES is not large enough");
+
+using namespace mozilla;
+using namespace mozilla::dom;
+using namespace mozilla::gfx;
+using namespace mozilla::layers;
+
+mozilla::LazyLogModule gMediaPipelineLog("MediaPipeline");
+
+namespace mozilla {
+
+// An async inserter for audio data, to avoid running audio codec encoders
+// on the MTG/input audio thread.  Basically just bounces all the audio
+// data to a single audio processing/input queue.  We could if we wanted to
+// use multiple threads and a TaskQueue.
+class AudioProxyThread {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioProxyThread)
+
+  explicit AudioProxyThread(RefPtr<AudioSessionConduit> aConduit)
+      : mConduit(std::move(aConduit)),
+        mTaskQueue(TaskQueue::Create(
+            GetMediaThreadPool(MediaThreadType::WEBRTC_WORKER), "AudioProxy")),
+        mAudioConverter(nullptr) {
+    MOZ_ASSERT(mConduit);
+    MOZ_COUNT_CTOR(AudioProxyThread);
+  }
+
+  // This function is the identity if aInputRate is supported.
+  // Else, it returns a rate that is supported, that ensure no loss in audio
+  // quality: the sampling rate returned is always greater to the inputed
+  // sampling-rate, if they differ..
+  uint32_t AppropriateSendingRateForInputRate(uint32_t aInputRate) {
+    AudioSessionConduit* conduit =
+        static_cast<AudioSessionConduit*>(mConduit.get());
+    if (conduit->IsSamplingFreqSupported(aInputRate)) {
+      return aInputRate;
+    }
+    if (aInputRate < 16000) {
+      return 16000;
+    }
+    if (aInputRate < 32000) {
+      return 32000;
+    }
+    if (aInputRate < 44100) {
+      return 44100;
+    }
+    return 48000;
+  }
+
+  // From an arbitrary AudioChunk at sampling-rate aRate, process the audio into
+  // something the conduit can work with (or send silence if the track is not
+  // enabled), and send the audio in 10ms chunks to the conduit.
+  void InternalProcessAudioChunk(TrackRate aRate, const AudioChunk& aChunk,
+                                 bool aEnabled) {
+    MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
+
+    // Convert to interleaved 16-bits integer audio, with a maximum of two
+    // channels (since the WebRTC.org code below makes the assumption that the
+    // input audio is either mono or stereo), with a sample-rate rate that is
+    // 16, 32, 44.1, or 48kHz.
+    uint32_t outputChannels = aChunk.ChannelCount() == 1 ? 1 : 2;
+    int32_t transmissionRate = AppropriateSendingRateForInputRate(aRate);
+
+    // We take advantage of the fact that the common case (microphone directly
+    // to PeerConnection, that is, a normal call), the samples are already
+    // 16-bits mono, so the representation in interleaved and planar is the
+    // same, and we can just use that.
+    if (aEnabled && outputChannels == 1 &&
+        aChunk.mBufferFormat == AUDIO_FORMAT_S16 && transmissionRate == aRate) {
+      const int16_t* samples = aChunk.ChannelData<int16_t>().Elements()[0];
+      PacketizeAndSend(samples, transmissionRate, outputChannels,
+                       aChunk.mDuration);
+      return;
+    }
+
+    uint32_t sampleCount = aChunk.mDuration * outputChannels;
+    if (mInterleavedAudio.Length() < sampleCount) {
+      mInterleavedAudio.SetLength(sampleCount);
+    }
+
+    if (!aEnabled || aChunk.mBufferFormat == AUDIO_FORMAT_SILENCE) {
+      PodZero(mInterleavedAudio.Elements(), sampleCount);
+    } else if (aChunk.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
+      DownmixAndInterleave(aChunk.ChannelData<float>(), aChunk.mDuration,
+                           aChunk.mVolume, outputChannels,
+                           mInterleavedAudio.Elements());
+    } else if (aChunk.mBufferFormat == AUDIO_FORMAT_S16) {
+      DownmixAndInterleave(aChunk.ChannelData<int16_t>(), aChunk.mDuration,
+                           aChunk.mVolume, outputChannels,
+                           mInterleavedAudio.Elements());
+    }
+    int16_t* inputAudio = mInterleavedAudio.Elements();
+    size_t inputAudioFrameCount = aChunk.mDuration;
+
+    AudioConfig inputConfig(AudioConfig::ChannelLayout(outputChannels), aRate,
+                            AudioConfig::FORMAT_S16);
+    AudioConfig outputConfig(AudioConfig::ChannelLayout(outputChannels),
+                             transmissionRate, AudioConfig::FORMAT_S16);
+    // Resample to an acceptable sample-rate for the sending side
+    if (!mAudioConverter || mAudioConverter->InputConfig() != inputConfig ||
+        mAudioConverter->OutputConfig() != outputConfig) {
+      mAudioConverter = MakeUnique<AudioConverter>(inputConfig, outputConfig);
+    }
+
+    int16_t* processedAudio = nullptr;
+    size_t framesProcessed =
+        mAudioConverter->Process(inputAudio, inputAudioFrameCount);
+
+    if (framesProcessed == 0) {
+      // In place conversion not possible, use a buffer.
+      framesProcessed = mAudioConverter->Process(mOutputAudio, inputAudio,
+                                                 inputAudioFrameCount);
+      processedAudio = mOutputAudio.Data();
+    } else {
+      processedAudio = inputAudio;
+    }
+
+    PacketizeAndSend(processedAudio, transmissionRate, outputChannels,
+                     framesProcessed);
+  }
+
+  // This packetizes aAudioData in 10ms chunks and sends it.
+  // aAudioData is interleaved audio data at a rate and with a channel count
+  // that is appropriate to send with the conduit.
+  void PacketizeAndSend(const int16_t* aAudioData, uint32_t aRate,
+                        uint32_t aChannels, uint32_t aFrameCount) {
+    MOZ_ASSERT(AppropriateSendingRateForInputRate(aRate) == aRate);
+    MOZ_ASSERT(aChannels == 1 || aChannels == 2);
+    MOZ_ASSERT(aAudioData);
+
+    uint32_t audio_10ms = aRate / 100;
+
+    if (!mPacketizer || mPacketizer->mPacketSize != audio_10ms ||
+        mPacketizer->mChannels != aChannels) {
+      // It's the right thing to drop the bit of audio still in the packetizer:
+      // we don't want to send to the conduit audio that has two different
+      // rates while telling it that it has a constante rate.
+      mPacketizer =
+          MakeUnique<AudioPacketizer<int16_t, int16_t>>(audio_10ms, aChannels);
+      mPacket = MakeUnique<int16_t[]>(audio_10ms * aChannels);
+    }
+
+    mPacketizer->Input(aAudioData, aFrameCount);
+
+    while (mPacketizer->PacketsAvailable()) {
+      mPacketizer->Output(mPacket.get());
+      auto frame = std::make_unique<webrtc::AudioFrame>();
+      // UpdateFrame makes a copy of the audio data.
+      frame->UpdateFrame(frame->timestamp_, mPacket.get(),
+                         mPacketizer->mPacketSize, aRate, frame->speech_type_,
+                         frame->vad_activity_, mPacketizer->mChannels);
+      mConduit->SendAudioFrame(std::move(frame));
+    }
+  }
+
+  void QueueAudioChunk(TrackRate aRate, const AudioChunk& aChunk,
+                       bool aEnabled) {
+    RefPtr<AudioProxyThread> self = this;
+    nsresult rv = mTaskQueue->Dispatch(NS_NewRunnableFunction(
+        "AudioProxyThread::QueueAudioChunk", [self, aRate, aChunk, aEnabled]() {
+          self->InternalProcessAudioChunk(aRate, aChunk, aEnabled);
+        }));
+    MOZ_DIAGNOSTIC_ASSERT(NS_SUCCEEDED(rv));
+    Unused << rv;
+  }
+
+ protected:
+  virtual ~AudioProxyThread() { MOZ_COUNT_DTOR(AudioProxyThread); }
+
+  const RefPtr<AudioSessionConduit> mConduit;
+  const RefPtr<TaskQueue> mTaskQueue;
+  // Only accessed on mTaskQueue
+  UniquePtr<AudioPacketizer<int16_t, int16_t>> mPacketizer;
+  // A buffer to hold a single packet of audio.
+  UniquePtr<int16_t[]> mPacket;
+  nsTArray<int16_t> mInterleavedAudio;
+  AlignedShortBuffer mOutputAudio;
+  UniquePtr<AudioConverter> mAudioConverter;
+};
+
+#define INIT_MIRROR(name, val) \
+  name(AbstractThread::MainThread(), val, "MediaPipeline::" #name " (Mirror)")
+
+MediaPipeline::MediaPipeline(const std::string& aPc,
+                             RefPtr<MediaTransportHandler> aTransportHandler,
+                             DirectionType aDirection,
+                             RefPtr<AbstractThread> aCallThread,
+                             RefPtr<nsISerialEventTarget> aStsThread,
+                             RefPtr<MediaSessionConduit> aConduit)
+    : mConduit(std::move(aConduit)),
+      mDirection(aDirection),
+      mCallThread(std::move(aCallThread)),
+      mStsThread(std::move(aStsThread)),
+      INIT_MIRROR(mActive, false),
+      mLevel(0),
+      mTransportHandler(std::move(aTransportHandler)),
+      mRtpPacketsSent(0),
+      mRtcpPacketsSent(0),
+      mRtpPacketsReceived(0),
+      mRtcpPacketsReceived(0),
+      mRtpBytesSent(0),
+      mRtpBytesReceived(0),
+      mPc(aPc),
+      mFilter(),
+      mRtpHeaderExtensionMap(new webrtc::RtpHeaderExtensionMap()),
+      mPacketDumper(PacketDumper::GetPacketDumper(mPc)) {}
+
+#undef INIT_MIRROR
+
+MediaPipeline::~MediaPipeline() {
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+          ("Destroying MediaPipeline: %s", mDescription.c_str()));
+}
+
+void MediaPipeline::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  mActive.DisconnectIfConnected();
+  RUN_ON_THREAD(mStsThread,
+                WrapRunnable(RefPtr<MediaPipeline>(this),
+                             &MediaPipeline::DetachTransport_s),
+                NS_DISPATCH_NORMAL);
+}
+
+void MediaPipeline::DetachTransport_s() {
+  ASSERT_ON_THREAD(mStsThread);
+
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+          ("%s in %s", mDescription.c_str(), __FUNCTION__));
+
+  disconnect_all();
+  mRtpState = TransportLayer::TS_NONE;
+  mRtcpState = TransportLayer::TS_NONE;
+  mTransportId.clear();
+  mConduit->SetTransportActive(false);
+  mRtpSendEventListener.DisconnectIfExists();
+  mSenderRtcpSendEventListener.DisconnectIfExists();
+  mReceiverRtcpSendEventListener.DisconnectIfExists();
+}
+
+void MediaPipeline::UpdateTransport_m(
+    const std::string& aTransportId, UniquePtr<MediaPipelineFilter>&& aFilter) {
+  mStsThread->Dispatch(NS_NewRunnableFunction(
+      __func__, [aTransportId, filter = std::move(aFilter),
+                 self = RefPtr<MediaPipeline>(this)]() mutable {
+        self->UpdateTransport_s(aTransportId, std::move(filter));
+      }));
+}
+
+void MediaPipeline::UpdateTransport_s(
+    const std::string& aTransportId, UniquePtr<MediaPipelineFilter>&& aFilter) {
+  ASSERT_ON_THREAD(mStsThread);
+  if (!mSignalsConnected) {
+    mTransportHandler->SignalStateChange.connect(
+        this, &MediaPipeline::RtpStateChange);
+    mTransportHandler->SignalRtcpStateChange.connect(
+        this, &MediaPipeline::RtcpStateChange);
+    mTransportHandler->SignalEncryptedSending.connect(
+        this, &MediaPipeline::EncryptedPacketSending);
+    mTransportHandler->SignalPacketReceived.connect(
+        this, &MediaPipeline::PacketReceived);
+    mTransportHandler->SignalAlpnNegotiated.connect(
+        this, &MediaPipeline::AlpnNegotiated);
+    mSignalsConnected = true;
+  }
+
+  if (aTransportId != mTransportId) {
+    mTransportId = aTransportId;
+    mRtpState = mTransportHandler->GetState(mTransportId, false);
+    mRtcpState = mTransportHandler->GetState(mTransportId, true);
+    CheckTransportStates();
+  }
+
+  if (mFilter) {
+    for (const auto& extension : mFilter->GetExtmap()) {
+      mRtpHeaderExtensionMap->Deregister(extension.uri);
+    }
+  }
+  if (mFilter && aFilter) {
+    // Use the new filter, but don't forget any remote SSRCs that we've learned
+    // by receiving traffic.
+    mFilter->Update(*aFilter);
+  } else {
+    mFilter = std::move(aFilter);
+  }
+  if (mFilter) {
+    for (const auto& extension : mFilter->GetExtmap()) {
+      mRtpHeaderExtensionMap->RegisterByUri(extension.id, extension.uri);
+    }
+  }
+}
+
+void MediaPipeline::GetContributingSourceStats(
+    const nsString& aInboundRtpStreamId,
+    FallibleTArray<dom::RTCRTPContributingSourceStats>& aArr) const {
+  ASSERT_ON_THREAD(mStsThread);
+  // Get the expiry from now
+  DOMHighResTimeStamp expiry =
+      RtpCSRCStats::GetExpiryFromTime(GetTimestampMaker().GetNow().ToDom());
+  for (auto info : mCsrcStats) {
+    if (!info.second.Expired(expiry)) {
+      RTCRTPContributingSourceStats stats;
+      info.second.GetWebidlInstance(stats, aInboundRtpStreamId);
+      if (!aArr.AppendElement(stats, fallible)) {
+        mozalloc_handle_oom(0);
+      }
+    }
+  }
+}
+
+void MediaPipeline::RtpStateChange(const std::string& aTransportId,
+                                   TransportLayer::State aState) {
+  if (mTransportId != aTransportId) {
+    return;
+  }
+  mRtpState = aState;
+  CheckTransportStates();
+}
+
+void MediaPipeline::RtcpStateChange(const std::string& aTransportId,
+                                    TransportLayer::State aState) {
+  if (mTransportId != aTransportId) {
+    return;
+  }
+  mRtcpState = aState;
+  CheckTransportStates();
+}
+
+void MediaPipeline::CheckTransportStates() {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (mRtpState == TransportLayer::TS_CLOSED ||
+      mRtpState == TransportLayer::TS_ERROR ||
+      mRtcpState == TransportLayer::TS_CLOSED ||
+      mRtcpState == TransportLayer::TS_ERROR) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Warning,
+            ("RTP Transport failed for pipeline %p flow %s", this,
+             mDescription.c_str()));
+
+    NS_WARNING(
+        "MediaPipeline Transport failed. This is not properly cleaned up yet");
+    // TODO(ekr@rtfm.com): SECURITY: Figure out how to clean up if the
+    // connection was good and now it is bad.
+    // TODO(ekr@rtfm.com): Report up so that the PC knows we
+    // have experienced an error.
+    mConduit->SetTransportActive(false);
+    mRtpSendEventListener.DisconnectIfExists();
+    mSenderRtcpSendEventListener.DisconnectIfExists();
+    mReceiverRtcpSendEventListener.DisconnectIfExists();
+    return;
+  }
+
+  if (mRtpState == TransportLayer::TS_OPEN) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTP Transport ready for pipeline %p flow %s", this,
+             mDescription.c_str()));
+  }
+
+  if (mRtcpState == TransportLayer::TS_OPEN) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTCP Transport ready for pipeline %p flow %s", this,
+             mDescription.c_str()));
+  }
+
+  if (mRtpState == TransportLayer::TS_OPEN && mRtcpState == mRtpState) {
+    if (mDirection == DirectionType::TRANSMIT) {
+      mConduit->ConnectSenderRtcpEvent(mSenderRtcpReceiveEvent);
+      mRtpSendEventListener = mConduit->SenderRtpSendEvent().Connect(
+          mStsThread, this, &MediaPipeline::SendPacket);
+      mSenderRtcpSendEventListener = mConduit->SenderRtcpSendEvent().Connect(
+          mStsThread, this, &MediaPipeline::SendPacket);
+    } else {
+      mConduit->ConnectReceiverRtcpEvent(mReceiverRtcpReceiveEvent);
+      mConduit->ConnectReceiverRtpEvent(mRtpReceiveEvent);
+      mReceiverRtcpSendEventListener =
+          mConduit->ReceiverRtcpSendEvent().Connect(mStsThread, this,
+                                                    &MediaPipeline::SendPacket);
+    }
+    mConduit->SetTransportActive(true);
+    TransportReady_s();
+  }
+}
+
+void MediaPipeline::SendPacket(MediaPacket&& aPacket) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  const bool isRtp = aPacket.type() == MediaPacket::RTP;
+
+  if (isRtp && mRtpState != TransportLayer::TS_OPEN) {
+    return;
+  }
+
+  if (!isRtp && mRtcpState != TransportLayer::TS_OPEN) {
+    return;
+  }
+
+  aPacket.sdp_level() = Some(Level());
+
+  if (RtpLogger::IsPacketLoggingOn()) {
+    RtpLogger::LogPacket(aPacket, false, mDescription);
+  }
+
+  if (isRtp) {
+    mPacketDumper->Dump(Level(), dom::mozPacketDumpType::Rtp, true,
+                        aPacket.data(), aPacket.len());
+    IncrementRtpPacketsSent(aPacket);
+  } else {
+    mPacketDumper->Dump(Level(), dom::mozPacketDumpType::Rtcp, true,
+                        aPacket.data(), aPacket.len());
+    IncrementRtcpPacketsSent();
+  }
+
+  MOZ_LOG(
+      gMediaPipelineLog, LogLevel::Debug,
+      ("%s sending %s packet", mDescription.c_str(), (isRtp ? "RTP" : "RTCP")));
+
+  mTransportHandler->SendPacket(mTransportId, std::move(aPacket));
+}
+
+void MediaPipeline::IncrementRtpPacketsSent(const MediaPacket& aPacket) {
+  ASSERT_ON_THREAD(mStsThread);
+  ++mRtpPacketsSent;
+  mRtpBytesSent += aPacket.len();
+
+  if (!(mRtpPacketsSent % 100)) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTP sent packet count for %s Pipeline %p: %u (%" PRId64 " bytes)",
+             mDescription.c_str(), this, mRtpPacketsSent, mRtpBytesSent));
+  }
+}
+
+void MediaPipeline::IncrementRtcpPacketsSent() {
+  ASSERT_ON_THREAD(mStsThread);
+  ++mRtcpPacketsSent;
+  if (!(mRtcpPacketsSent % 100)) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTCP sent packet count for %s Pipeline %p: %u",
+             mDescription.c_str(), this, mRtcpPacketsSent));
+  }
+}
+
+void MediaPipeline::IncrementRtpPacketsReceived(int32_t aBytes) {
+  ASSERT_ON_THREAD(mStsThread);
+  ++mRtpPacketsReceived;
+  mRtpBytesReceived += aBytes;
+  if (!(mRtpPacketsReceived % 100)) {
+    MOZ_LOG(
+        gMediaPipelineLog, LogLevel::Info,
+        ("RTP received packet count for %s Pipeline %p: %u (%" PRId64 " bytes)",
+         mDescription.c_str(), this, mRtpPacketsReceived, mRtpBytesReceived));
+  }
+}
+
+void MediaPipeline::IncrementRtcpPacketsReceived() {
+  ASSERT_ON_THREAD(mStsThread);
+  ++mRtcpPacketsReceived;
+  if (!(mRtcpPacketsReceived % 100)) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTCP received packet count for %s Pipeline %p: %u",
+             mDescription.c_str(), this, mRtcpPacketsReceived));
+  }
+}
+
+void MediaPipeline::RtpPacketReceived(const MediaPacket& packet) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (mDirection == DirectionType::TRANSMIT) {
+    return;
+  }
+
+  if (!packet.len()) {
+    return;
+  }
+
+  webrtc::RTPHeader header;
+  rtc::CopyOnWriteBuffer packet_buffer(packet.data(), packet.len());
+  webrtc::RtpPacketReceived parsedPacket(mRtpHeaderExtensionMap.get());
+  if (!parsedPacket.Parse(packet_buffer)) {
+    return;
+  }
+  parsedPacket.GetHeader(&header);
+
+  if (mFilter && !mFilter->Filter(header)) {
+    return;
+  }
+
+  auto now = GetTimestampMaker().GetNow();
+  parsedPacket.set_arrival_time(now.ToRealtime());
+  if (IsVideo()) {
+    parsedPacket.set_payload_type_frequency(webrtc::kVideoPayloadTypeFrequency);
+  }
+
+  // Remove expired RtpCSRCStats
+  if (!mCsrcStats.empty()) {
+    auto expiry = RtpCSRCStats::GetExpiryFromTime(now.ToDom());
+    for (auto p = mCsrcStats.begin(); p != mCsrcStats.end();) {
+      if (p->second.Expired(expiry)) {
+        p = mCsrcStats.erase(p);
+        continue;
+      }
+      p++;
+    }
+  }
+
+  // Add new RtpCSRCStats
+  if (header.numCSRCs) {
+    for (auto i = 0; i < header.numCSRCs; i++) {
+      auto csrcInfo = mCsrcStats.find(header.arrOfCSRCs[i]);
+      if (csrcInfo == mCsrcStats.end()) {
+        mCsrcStats.insert(
+            std::make_pair(header.arrOfCSRCs[i],
+                           RtpCSRCStats(header.arrOfCSRCs[i], now.ToDom())));
+      } else {
+        csrcInfo->second.SetTimestamp(now.ToDom());
+      }
+    }
+  }
+
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+          ("%s received RTP packet.", mDescription.c_str()));
+  IncrementRtpPacketsReceived(packet.len());
+
+  RtpLogger::LogPacket(packet, true, mDescription);
+
+  // Might be nice to pass ownership of the buffer in this case, but it is a
+  // small optimization in a rare case.
+  mPacketDumper->Dump(mLevel, dom::mozPacketDumpType::Srtp, false,
+                      packet.encrypted_data(), packet.encrypted_len());
+
+  mPacketDumper->Dump(mLevel, dom::mozPacketDumpType::Rtp, false, packet.data(),
+                      packet.len());
+
+  mRtpReceiveEvent.Notify(std::move(parsedPacket), header);
+}
+
+void MediaPipeline::RtcpPacketReceived(const MediaPacket& packet) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (!packet.len()) {
+    return;
+  }
+
+  // We do not filter RTCP. This is because a compound RTCP packet can contain
+  // any collection of RTCP packets, and webrtc.org already knows how to filter
+  // out what it is interested in, and what it is not. Maybe someday we should
+  // have a TransportLayer that breaks up compound RTCP so we can filter them
+  // individually, but I doubt that will matter much.
+
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+          ("%s received RTCP packet.", mDescription.c_str()));
+  IncrementRtcpPacketsReceived();
+
+  RtpLogger::LogPacket(packet, true, mDescription);
+
+  // Might be nice to pass ownership of the buffer in this case, but it is a
+  // small optimization in a rare case.
+  mPacketDumper->Dump(mLevel, dom::mozPacketDumpType::Srtcp, false,
+                      packet.encrypted_data(), packet.encrypted_len());
+
+  mPacketDumper->Dump(mLevel, dom::mozPacketDumpType::Rtcp, false,
+                      packet.data(), packet.len());
+
+  if (StaticPrefs::media_webrtc_net_force_disable_rtcp_reception()) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+            ("%s RTCP packet forced to be dropped", mDescription.c_str()));
+    return;
+  }
+
+  if (mDirection == DirectionType::TRANSMIT) {
+    mSenderRtcpReceiveEvent.Notify(packet.Clone());
+  } else {
+    mReceiverRtcpReceiveEvent.Notify(packet.Clone());
+  }
+}
+
+void MediaPipeline::PacketReceived(const std::string& aTransportId,
+                                   const MediaPacket& packet) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (mTransportId != aTransportId) {
+    return;
+  }
+
+  MOZ_ASSERT(mRtpState == TransportLayer::TS_OPEN);
+  MOZ_ASSERT(mRtcpState == mRtpState);
+
+  switch (packet.type()) {
+    case MediaPacket::RTP:
+      RtpPacketReceived(packet);
+      break;
+    case MediaPacket::RTCP:
+      RtcpPacketReceived(packet);
+      break;
+    default:;
+  }
+}
+
+void MediaPipeline::AlpnNegotiated(const std::string& aAlpn,
+                                   bool aPrivacyRequested) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (aPrivacyRequested && Direction() == DirectionType::RECEIVE) {
+    // This will force the receive pipelines to drop data until they have
+    // received a private PrincipalHandle from RTCRtpReceiver (which takes a
+    // detour via main thread).
+    static_cast<MediaPipelineReceive*>(this)->OnPrivacyRequested_s();
+  }
+}
+
+void MediaPipeline::EncryptedPacketSending(const std::string& aTransportId,
+                                           const MediaPacket& aPacket) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (mTransportId == aTransportId) {
+    dom::mozPacketDumpType type;
+    if (aPacket.type() == MediaPacket::SRTP) {
+      type = dom::mozPacketDumpType::Srtp;
+    } else if (aPacket.type() == MediaPacket::SRTCP) {
+      type = dom::mozPacketDumpType::Srtcp;
+    } else if (aPacket.type() == MediaPacket::DTLS) {
+      // TODO(bug 1497936): Implement packet dump for DTLS
+      return;
+    } else {
+      MOZ_ASSERT(false);
+      return;
+    }
+    mPacketDumper->Dump(Level(), type, true, aPacket.data(), aPacket.len());
+  }
+}
+
+class MediaPipelineTransmit::PipelineListener
+    : public DirectMediaTrackListener {
+  friend class MediaPipelineTransmit;
+
+ public:
+  explicit PipelineListener(RefPtr<MediaSessionConduit> aConduit)
+      : mConduit(std::move(aConduit)),
+        mActive(false),
+        mEnabled(false),
+        mDirectConnect(false) {}
+
+  ~PipelineListener() {
+    if (mConverter) {
+      mConverter->Shutdown();
+    }
+  }
+
+  void SetActive(bool aActive) {
+    mActive = aActive;
+    if (mConverter) {
+      mConverter->SetActive(aActive);
+    }
+  }
+  void SetEnabled(bool aEnabled) { mEnabled = aEnabled; }
+
+  // These are needed since nested classes don't have access to any particular
+  // instance of the parent
+  void SetAudioProxy(RefPtr<AudioProxyThread> aProxy) {
+    mAudioProcessing = std::move(aProxy);
+  }
+
+  void SetVideoFrameConverter(RefPtr<VideoFrameConverter> aConverter) {
+    mConverter = std::move(aConverter);
+  }
+
+  void OnVideoFrameConverted(webrtc::VideoFrame aVideoFrame) {
+    MOZ_RELEASE_ASSERT(mConduit->type() == MediaSessionConduit::VIDEO);
+    static_cast<VideoSessionConduit*>(mConduit.get())
+        ->SendVideoFrame(std::move(aVideoFrame));
+  }
+
+  // Implement MediaTrackListener
+  void NotifyQueuedChanges(MediaTrackGraph* aGraph, TrackTime aOffset,
+                           const MediaSegment& aQueuedMedia) override;
+  void NotifyEnabledStateChanged(MediaTrackGraph* aGraph,
+                                 bool aEnabled) override;
+
+  // Implement DirectMediaTrackListener
+  void NotifyRealtimeTrackData(MediaTrackGraph* aGraph, TrackTime aOffset,
+                               const MediaSegment& aMedia) override;
+  void NotifyDirectListenerInstalled(InstallationResult aResult) override;
+  void NotifyDirectListenerUninstalled() override;
+
+ private:
+  void NewData(const MediaSegment& aMedia, TrackRate aRate = 0);
+
+  const RefPtr<MediaSessionConduit> mConduit;
+  RefPtr<AudioProxyThread> mAudioProcessing;
+  RefPtr<VideoFrameConverter> mConverter;
+
+  // active is true if there is a transport to send on
+  mozilla::Atomic<bool> mActive;
+  // enabled is true if the media access control permits sending
+  // actual content; when false you get black/silence
+  mozilla::Atomic<bool> mEnabled;
+
+  // Written and read on the MediaTrackGraph thread
+  bool mDirectConnect;
+};
+
+MediaPipelineTransmit::MediaPipelineTransmit(
+    const std::string& aPc, RefPtr<MediaTransportHandler> aTransportHandler,
+    RefPtr<AbstractThread> aCallThread, RefPtr<nsISerialEventTarget> aStsThread,
+    bool aIsVideo, RefPtr<MediaSessionConduit> aConduit)
+    : MediaPipeline(aPc, std::move(aTransportHandler), DirectionType::TRANSMIT,
+                    std::move(aCallThread), std::move(aStsThread),
+                    std::move(aConduit)),
+      mWatchManager(this, AbstractThread::MainThread()),
+      mIsVideo(aIsVideo),
+      mListener(new PipelineListener(mConduit)),
+      mDomTrack(nullptr, "MediaPipelineTransmit::mDomTrack"),
+      mSendTrackOverride(nullptr, "MediaPipelineTransmit::mSendTrackOverride") {
+  if (!IsVideo()) {
+    mAudioProcessing =
+        MakeAndAddRef<AudioProxyThread>(*mConduit->AsAudioSessionConduit());
+    mListener->SetAudioProxy(mAudioProcessing);
+  } else {  // Video
+    mConverter = MakeAndAddRef<VideoFrameConverter>(GetTimestampMaker());
+    mFrameListener = mConverter->VideoFrameConvertedEvent().Connect(
+        mConverter->mTaskQueue,
+        [listener = mListener](webrtc::VideoFrame aFrame) {
+          listener->OnVideoFrameConverted(std::move(aFrame));
+        });
+    mListener->SetVideoFrameConverter(mConverter);
+  }
+
+  mWatchManager.Watch(mActive, &MediaPipelineTransmit::UpdateSendState);
+  mWatchManager.Watch(mDomTrack, &MediaPipelineTransmit::UpdateSendState);
+  mWatchManager.Watch(mSendTrackOverride,
+                      &MediaPipelineTransmit::UpdateSendState);
+
+  mDescription = GenerateDescription();
+}
+
+MediaPipelineTransmit::~MediaPipelineTransmit() {
+  mFrameListener.DisconnectIfExists();
+
+  MOZ_ASSERT(!mTransmitting);
+  MOZ_ASSERT(!mDomTrack.Ref());
+}
+
+void MediaPipelineTransmit::InitControl(
+    MediaPipelineTransmitControlInterface* aControl) {
+  mActive.Connect(aControl->CanonicalTransmitting());
+}
+
+void MediaPipelineTransmit::Shutdown() {
+  MediaPipeline::Shutdown();
+  mWatchManager.Shutdown();
+  if (mDomTrack.Ref()) {
+    mDomTrack.Ref()->RemovePrincipalChangeObserver(this);
+    mDomTrack = nullptr;
+  }
+  mUnsettingSendTrack = false;
+  UpdateSendState();
+  MOZ_ASSERT(!mTransmitting);
+}
+
+void MediaPipeline::SetDescription_s(const std::string& description) {
+  ASSERT_ON_THREAD(mStsThread);
+  mDescription = description;
+}
+
+std::string MediaPipelineTransmit::GenerateDescription() const {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  std::stringstream description;
+  description << mPc << "| ";
+  description << (mIsVideo ? "Transmit video[" : "Transmit audio[");
+
+  if (mDomTrack.Ref()) {
+    nsString nsTrackId;
+    mDomTrack.Ref()->GetId(nsTrackId);
+    description << NS_ConvertUTF16toUTF8(nsTrackId).get();
+  } else if (mSendTrackOverride.Ref()) {
+    description << "override " << mSendTrackOverride.Ref().get();
+  } else {
+    description << "no track";
+  }
+
+  description << "]";
+
+  return description.str();
+}
+
+void MediaPipelineTransmit::UpdateSendState() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  // This runs because either mActive, mDomTrack or mSendTrackOverride changed,
+  // or because mSendTrack was unset async. Based on these inputs this method
+  // is responsible for hooking up mSendTrack to mListener in order to feed data
+  // to the conduit.
+  //
+  // If we are inactive, or if the send track does not match what we want to
+  // send (mDomTrack or mSendTrackOverride), we must stop feeding data to the
+  // conduit. NB that removing the listener from mSendTrack is async, and we
+  // must wait for it to resolve before adding mListener to another track.
+  // mUnsettingSendTrack gates us until the listener has been removed from
+  // mSendTrack.
+  //
+  // If we are active and the send track does match what we want to send, we
+  // make sure mListener is added to the send track. Either now, or if we're
+  // still waiting for another send track to be removed, during a future call to
+  // this method.
+
+  if (mUnsettingSendTrack) {
+    // We must wait for the send track to be unset before we can set it again,
+    // to avoid races. Once unset this function is triggered again.
+    return;
+  }
+
+  const bool wasTransmitting = mTransmitting;
+
+  const bool haveLiveSendTrack = mSendTrack && !mSendTrack->IsDestroyed();
+  const bool haveLiveDomTrack = mDomTrack.Ref() && !mDomTrack.Ref()->Ended();
+  const bool haveLiveOverrideTrack =
+      mSendTrackOverride.Ref() && !mSendTrackOverride.Ref()->IsDestroyed();
+  const bool mustRemoveSendTrack =
+      haveLiveSendTrack && !mSendTrackOverride.Ref() &&
+      (!haveLiveDomTrack || mDomTrack.Ref()->GetTrack() != mSendPortSource);
+
+  mTransmitting = mActive && (haveLiveDomTrack || haveLiveOverrideTrack) &&
+                  !mustRemoveSendTrack;
+
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+          ("MediaPipeline %p UpdateSendState wasTransmitting=%d, active=%d, "
+           "sendTrack=%p (%s), domTrack=%p (%s), "
+           "sendTrackOverride=%p (%s), mustRemove=%d, mTransmitting=%d",
+           this, wasTransmitting, mActive.Ref(), mSendTrack.get(),
+           haveLiveSendTrack ? "live" : "ended", mDomTrack.Ref().get(),
+           haveLiveDomTrack ? "live" : "ended", mSendTrackOverride.Ref().get(),
+           haveLiveOverrideTrack ? "live" : "ended", mustRemoveSendTrack,
+           mTransmitting));
+
+  if (!wasTransmitting && mTransmitting) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+            ("Attaching pipeline %p to track %p conduit type=%s", this,
+             mDomTrack.Ref().get(), mIsVideo ? "video" : "audio"));
+    if (mDescriptionInvalidated) {
+      // Only update the description when we attach to a track, as detaching is
+      // always a longer async step than updating the description. Updating on
+      // detach would cause the wrong track id to be attributed in logs.
+      RUN_ON_THREAD(mStsThread,
+                    WrapRunnable(RefPtr<MediaPipeline>(this),
+                                 &MediaPipelineTransmit::SetDescription_s,
+                                 GenerateDescription()),
+                    NS_DISPATCH_NORMAL);
+      mDescriptionInvalidated = false;
+    }
+    if (mSendTrackOverride.Ref()) {
+      // Special path that allows unittests to avoid mDomTrack and the graph by
+      // manually calling SetSendTrack.
+      mSendTrack = mSendTrackOverride.Ref();
+    } else {
+      mSendTrack = mDomTrack.Ref()->Graph()->CreateForwardedInputTrack(
+          mDomTrack.Ref()->GetTrack()->mType);
+      mSendPortSource = mDomTrack.Ref()->GetTrack();
+      mSendPort = mSendTrack->AllocateInputPort(mSendPortSource.get());
+    }
+    if (mIsVideo) {
+      mConverter->SetTrackingId(mDomTrack.Ref()->GetSource().mTrackingId);
+    }
+    mSendTrack->QueueSetAutoend(false);
+    if (mIsVideo) {
+      mSendTrack->AddDirectListener(mListener);
+    }
+    mSendTrack->AddListener(mListener);
+  }
+
+  if (wasTransmitting && !mTransmitting) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+            ("Detaching pipeline %p from track %p conduit type=%s", this,
+             mDomTrack.Ref().get(), mIsVideo ? "video" : "audio"));
+    mUnsettingSendTrack = true;
+    if (mIsVideo) {
+      mSendTrack->RemoveDirectListener(mListener);
+    }
+    mSendTrack->RemoveListener(mListener)->Then(
+        GetMainThreadSerialEventTarget(), __func__,
+        [this, self = RefPtr<MediaPipelineTransmit>(this)] {
+          mUnsettingSendTrack = false;
+          mSendTrack = nullptr;
+          if (!mWatchManager.IsShutdown()) {
+            mWatchManager.ManualNotify(&MediaPipelineTransmit::UpdateSendState);
+          }
+        });
+    if (!mSendTrackOverride.Ref()) {
+      // If an override is set it may be re-used.
+      mSendTrack->Destroy();
+      mSendPort->Destroy();
+      mSendPort = nullptr;
+      mSendPortSource = nullptr;
+    }
+  }
+}
+
+bool MediaPipelineTransmit::Transmitting() const {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  return mActive;
+}
+
+bool MediaPipelineTransmit::IsVideo() const { return mIsVideo; }
+
+void MediaPipelineTransmit::PrincipalChanged(dom::MediaStreamTrack* aTrack) {
+  MOZ_ASSERT(aTrack && aTrack == mDomTrack.Ref());
+
+  PeerConnectionWrapper pcw(mPc);
+  if (pcw.impl()) {
+    Document* doc = pcw.impl()->GetParentObject()->GetExtantDoc();
+    if (doc) {
+      UpdateSinkIdentity(doc->NodePrincipal(), pcw.impl()->GetPeerIdentity());
+    } else {
+      MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+              ("Can't update sink principal; document gone"));
+    }
+  }
+}
+
+void MediaPipelineTransmit::UpdateSinkIdentity(
+    nsIPrincipal* aPrincipal, const PeerIdentity* aSinkIdentity) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (!mDomTrack.Ref()) {
+    // Nothing to do here
+    return;
+  }
+
+  bool enableTrack = aPrincipal->Subsumes(mDomTrack.Ref()->GetPrincipal());
+  if (!enableTrack) {
+    // first try didn't work, but there's a chance that this is still available
+    // if our track is bound to a peerIdentity, and the peer connection (our
+    // sink) is bound to the same identity, then we can enable the track.
+    const PeerIdentity* trackIdentity = mDomTrack.Ref()->GetPeerIdentity();
+    if (aSinkIdentity && trackIdentity) {
+      enableTrack = (*aSinkIdentity == *trackIdentity);
+    }
+  }
+
+  mListener->SetEnabled(enableTrack);
+}
+
+void MediaPipelineTransmit::TransportReady_s() {
+  ASSERT_ON_THREAD(mStsThread);
+  // Call base ready function.
+  MediaPipeline::TransportReady_s();
+  mListener->SetActive(true);
+}
+
+nsresult MediaPipelineTransmit::SetTrack(
+    const RefPtr<MediaStreamTrack>& aDomTrack) {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mDomTrack.Ref()) {
+    mDomTrack.Ref()->RemovePrincipalChangeObserver(this);
+  }
+
+  if (aDomTrack) {
+    nsString nsTrackId;
+    aDomTrack->GetId(nsTrackId);
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+            ("Reattaching pipeline to track %p track %s conduit type: %s",
+             aDomTrack.get(), NS_ConvertUTF16toUTF8(nsTrackId).get(),
+             mIsVideo ? "video" : "audio"));
+  }
+
+  mDescriptionInvalidated = true;
+  mDomTrack = aDomTrack;
+  if (mDomTrack.Ref()) {
+    mDomTrack.Ref()->AddPrincipalChangeObserver(this);
+    PrincipalChanged(mDomTrack.Ref());
+  }
+
+  return NS_OK;
+}
+
+RefPtr<dom::MediaStreamTrack> MediaPipelineTransmit::GetTrack() const {
+  MOZ_ASSERT(NS_IsMainThread());
+  return mDomTrack;
+}
+
+void MediaPipelineTransmit::SetSendTrackOverride(
+    const RefPtr<ProcessedMediaTrack>& aSendTrack) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_RELEASE_ASSERT(!mSendTrack);
+  MOZ_RELEASE_ASSERT(!mSendPort);
+  MOZ_RELEASE_ASSERT(!mSendTrackOverride.Ref());
+  mDescriptionInvalidated = true;
+  mSendTrackOverride = aSendTrack;
+}
+
+// Called if we're attached with AddDirectListener()
+void MediaPipelineTransmit::PipelineListener::NotifyRealtimeTrackData(
+    MediaTrackGraph* aGraph, TrackTime aOffset, const MediaSegment& aMedia) {
+  MOZ_LOG(
+      gMediaPipelineLog, LogLevel::Debug,
+      ("MediaPipeline::NotifyRealtimeTrackData() listener=%p, offset=%" PRId64
+       ", duration=%" PRId64,
+       this, aOffset, aMedia.GetDuration()));
+  TRACE_COMMENT(
+      "MediaPipelineTransmit::PipelineListener::NotifyRealtimeTrackData", "%s",
+      aMedia.GetType() == MediaSegment::VIDEO ? "Video" : "Audio");
+  NewData(aMedia, aGraph->GraphRate());
+}
+
+void MediaPipelineTransmit::PipelineListener::NotifyQueuedChanges(
+    MediaTrackGraph* aGraph, TrackTime aOffset,
+    const MediaSegment& aQueuedMedia) {
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+          ("MediaPipeline::NotifyQueuedChanges()"));
+
+  if (aQueuedMedia.GetType() == MediaSegment::VIDEO) {
+    // We always get video from the direct listener.
+    return;
+  }
+
+  TRACE("MediaPipelineTransmit::PipelineListener::NotifyQueuedChanges (Audio)");
+
+  if (mDirectConnect) {
+    // ignore non-direct data if we're also getting direct data
+    return;
+  }
+
+  size_t rate;
+  if (aGraph) {
+    rate = aGraph->GraphRate();
+  } else {
+    // When running tests, graph may be null. In that case use a default.
+    rate = 16000;
+  }
+  NewData(aQueuedMedia, rate);
+}
+
+void MediaPipelineTransmit::PipelineListener::NotifyEnabledStateChanged(
+    MediaTrackGraph* aGraph, bool aEnabled) {
+  if (mConduit->type() != MediaSessionConduit::VIDEO) {
+    return;
+  }
+  MOZ_ASSERT(mConverter);
+  mConverter->SetTrackEnabled(aEnabled);
+}
+
+void MediaPipelineTransmit::PipelineListener::NotifyDirectListenerInstalled(
+    InstallationResult aResult) {
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+          ("MediaPipeline::NotifyDirectListenerInstalled() listener=%p,"
+           " result=%d",
+           this, static_cast<int32_t>(aResult)));
+
+  mDirectConnect = InstallationResult::SUCCESS == aResult;
+}
+
+void MediaPipelineTransmit::PipelineListener::
+    NotifyDirectListenerUninstalled() {
+  MOZ_LOG(
+      gMediaPipelineLog, LogLevel::Info,
+      ("MediaPipeline::NotifyDirectListenerUninstalled() listener=%p", this));
+
+  if (mConduit->type() == MediaSessionConduit::VIDEO) {
+    // Reset the converter's track-enabled state. If re-added to a new track
+    // later and that track is disabled, we will be signaled explicitly.
+    MOZ_ASSERT(mConverter);
+    mConverter->SetTrackEnabled(true);
+  }
+
+  mDirectConnect = false;
+}
+
+void MediaPipelineTransmit::PipelineListener::NewData(
+    const MediaSegment& aMedia, TrackRate aRate /* = 0 */) {
+  if (mConduit->type() != (aMedia.GetType() == MediaSegment::AUDIO
+                               ? MediaSessionConduit::AUDIO
+                               : MediaSessionConduit::VIDEO)) {
+    MOZ_ASSERT(false,
+               "The media type should always be correct since the "
+               "listener is locked to a specific track");
+    return;
+  }
+
+  // TODO(ekr@rtfm.com): For now assume that we have only one
+  // track type and it's destined for us
+  // See bug 784517
+  if (aMedia.GetType() == MediaSegment::AUDIO) {
+    MOZ_RELEASE_ASSERT(aRate > 0);
+
+    if (!mActive) {
+      MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+              ("Discarding audio packets because transport not ready"));
+      return;
+    }
+
+    const AudioSegment* audio = static_cast<const AudioSegment*>(&aMedia);
+    for (AudioSegment::ConstChunkIterator iter(*audio); !iter.IsEnded();
+         iter.Next()) {
+      mAudioProcessing->QueueAudioChunk(aRate, *iter, mEnabled);
+    }
+  } else {
+    const VideoSegment* video = static_cast<const VideoSegment*>(&aMedia);
+
+    for (VideoSegment::ConstChunkIterator iter(*video); !iter.IsEnded();
+         iter.Next()) {
+      mConverter->QueueVideoChunk(*iter, !mEnabled);
+    }
+  }
+}
+
+class GenericReceiveListener : public MediaTrackListener {
+ public:
+  GenericReceiveListener(RefPtr<SourceMediaTrack> aSource,
+                         TrackingId aTrackingId)
+      : mSource(std::move(aSource)),
+        mTrackingId(std::move(aTrackingId)),
+        mIsAudio(mSource->mType == MediaSegment::AUDIO),
+        mEnabled(false) {
+    MOZ_DIAGNOSTIC_ASSERT(NS_IsMainThread());
+    MOZ_DIAGNOSTIC_ASSERT(mSource, "Must be used with a SourceMediaTrack");
+  }
+
+  virtual ~GenericReceiveListener() = default;
+
+  void Init() { mSource->AddListener(this); }
+  void Shutdown() { mSource->RemoveListener(this); }
+
+  void SetEnabled(bool aEnabled) {
+    if (mEnabled == aEnabled) {
+      return;
+    }
+    mEnabled = aEnabled;
+    if (mIsAudio && !mSource->IsDestroyed()) {
+      mSource->SetPullingEnabled(mEnabled);
+    }
+  }
+
+ protected:
+  const RefPtr<SourceMediaTrack> mSource;
+  const TrackingId mTrackingId;
+  const bool mIsAudio;
+  // Main thread only.
+  bool mEnabled;
+};
+
+MediaPipelineReceive::MediaPipelineReceive(
+    const std::string& aPc, RefPtr<MediaTransportHandler> aTransportHandler,
+    RefPtr<AbstractThread> aCallThread, RefPtr<nsISerialEventTarget> aStsThread,
+    RefPtr<MediaSessionConduit> aConduit)
+    : MediaPipeline(aPc, std::move(aTransportHandler), DirectionType::RECEIVE,
+                    std::move(aCallThread), std::move(aStsThread),
+                    std::move(aConduit)),
+      mWatchManager(this, AbstractThread::MainThread()) {
+  mWatchManager.Watch(mActive, &MediaPipelineReceive::UpdateListener);
+}
+
+MediaPipelineReceive::~MediaPipelineReceive() = default;
+
+void MediaPipelineReceive::InitControl(
+    MediaPipelineReceiveControlInterface* aControl) {
+  mActive.Connect(aControl->CanonicalReceiving());
+}
+
+void MediaPipelineReceive::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MediaPipeline::Shutdown();
+  mWatchManager.Shutdown();
+}
+
+class MediaPipelineReceiveAudio::PipelineListener
+    : public GenericReceiveListener {
+ public:
+  PipelineListener(RefPtr<SourceMediaTrack> aSource, TrackingId aTrackingId,
+                   RefPtr<MediaSessionConduit> aConduit,
+                   PrincipalHandle aPrincipalHandle, PrincipalPrivacy aPrivacy)
+      : GenericReceiveListener(std::move(aSource), std::move(aTrackingId)),
+        mConduit(std::move(aConduit)),
+        // AudioSession conduit only supports 16, 32, 44.1 and 48kHz
+        // This is an artificial limitation, it would however require more
+        // changes to support any rates. If the sampling rate is not-supported,
+        // we will use 48kHz instead.
+        mRate(static_cast<AudioSessionConduit*>(mConduit.get())
+                      ->IsSamplingFreqSupported(mSource->Graph()->GraphRate())
+                  ? mSource->Graph()->GraphRate()
+                  : WEBRTC_MAX_SAMPLE_RATE),
+        mTaskQueue(TaskQueue::Create(
+            GetMediaThreadPool(MediaThreadType::WEBRTC_WORKER),
+            "AudioPipelineListener")),
+        mPlayedTicks(0),
+        mAudioFrame(std::make_unique<webrtc::AudioFrame>()),
+        mPrincipalHandle(std::move(aPrincipalHandle)),
+        mPrivacy(aPrivacy),
+        mForceSilence(false) {}
+
+  void Init() {
+    GenericReceiveListener::Init();
+    mSource->SetAppendDataSourceRate(mRate);
+  }
+
+  // Implement MediaTrackListener
+  void NotifyPull(MediaTrackGraph* aGraph, TrackTime aEndOfAppendedData,
+                  TrackTime aDesiredTime) override {
+    NotifyPullImpl(aDesiredTime);
+  }
+
+  void OnPrivacyRequested_s() {
+    if (mPrivacy == PrincipalPrivacy::Private) {
+      return;
+    }
+    mForceSilence = true;
+  }
+
+  void SetPrivatePrincipal(PrincipalHandle aHandle) {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    class Message : public ControlMessage {
+     public:
+      Message(RefPtr<PipelineListener> aListener,
+              PrincipalHandle aPrivatePrincipal)
+          : ControlMessage(nullptr),
+            mListener(std::move(aListener)),
+            mPrivatePrincipal(std::move(aPrivatePrincipal)) {}
+
+      void Run() override {
+        if (mListener->mPrivacy == PrincipalPrivacy::Private) {
+          return;
+        }
+        mListener->mPrincipalHandle = mPrivatePrincipal;
+        mListener->mPrivacy = PrincipalPrivacy::Private;
+        mListener->mForceSilence = false;
+      }
+
+      const RefPtr<PipelineListener> mListener;
+      PrincipalHandle mPrivatePrincipal;
+    };
+
+    if (mSource->IsDestroyed()) {
+      return;
+    }
+
+    mSource->GraphImpl()->AppendMessage(
+        MakeUnique<Message>(this, std::move(aHandle)));
+  }
+
+ private:
+  ~PipelineListener() = default;
+
+  void NotifyPullImpl(TrackTime aDesiredTime) {
+    TRACE_COMMENT("PiplineListener::NotifyPullImpl", "PipelineListener %p",
+                  this);
+    uint32_t samplesPer10ms = mRate / 100;
+
+    // mSource's rate is not necessarily the same as the graph rate, since there
+    // are sample-rate constraints on the inbound audio: only 16, 32, 44.1 and
+    // 48kHz are supported. The audio frames we get here is going to be
+    // resampled when inserted into the graph. aDesiredTime and mPlayedTicks are
+    // in the graph rate.
+
+    while (mPlayedTicks < aDesiredTime) {
+      // This fetches 10ms of data, either mono or stereo
+      MediaConduitErrorCode err =
+          static_cast<AudioSessionConduit*>(mConduit.get())
+              ->GetAudioFrame(mRate, mAudioFrame.get());
+
+      if (err != kMediaConduitNoError) {
+        // Insert silence on conduit/GIPS failure (extremely unlikely)
+        MOZ_LOG(gMediaPipelineLog, LogLevel::Error,
+                ("Audio conduit failed (%d) to return data @ %" PRId64
+                 " (desired %" PRId64 " -> %f)",
+                 err, mPlayedTicks, aDesiredTime,
+                 mSource->TrackTimeToSeconds(aDesiredTime)));
+        constexpr size_t mono = 1;
+        mAudioFrame->UpdateFrame(
+            mAudioFrame->timestamp_, nullptr, samplesPer10ms, mRate,
+            mAudioFrame->speech_type_, mAudioFrame->vad_activity_,
+            std::max(mono, mAudioFrame->num_channels()));
+      }
+
+      MOZ_LOG(
+          gMediaPipelineLog, LogLevel::Debug,
+          ("Audio conduit returned buffer for %zu channels, %zu frames",
+           mAudioFrame->num_channels(), mAudioFrame->samples_per_channel()));
+
+      AudioSegment segment;
+      if (mForceSilence || mAudioFrame->muted()) {
+        segment.AppendNullData(mAudioFrame->samples_per_channel());
+      } else {
+        CheckedInt<size_t> bufferSize(sizeof(uint16_t));
+        bufferSize *= mAudioFrame->samples_per_channel();
+        bufferSize *= mAudioFrame->num_channels();
+        RefPtr<SharedBuffer> samples = SharedBuffer::Create(bufferSize);
+        int16_t* samplesData = static_cast<int16_t*>(samples->Data());
+        AutoTArray<int16_t*, 2> channels;
+        AutoTArray<const int16_t*, 2> outputChannels;
+
+        channels.SetLength(mAudioFrame->num_channels());
+
+        size_t offset = 0;
+        for (size_t i = 0; i < mAudioFrame->num_channels(); i++) {
+          channels[i] = samplesData + offset;
+          offset += mAudioFrame->samples_per_channel();
+        }
+
+        DeinterleaveAndConvertBuffer(
+            mAudioFrame->data(), mAudioFrame->samples_per_channel(),
+            mAudioFrame->num_channels(), channels.Elements());
+
+        outputChannels.AppendElements(channels);
+
+        segment.AppendFrames(samples.forget(), outputChannels,
+                             mAudioFrame->samples_per_channel(),
+                             mPrincipalHandle);
+      }
+
+      // Handle track not actually added yet or removed/finished
+      if (TrackTime appended = mSource->AppendData(&segment)) {
+        mPlayedTicks += appended;
+      } else {
+        MOZ_LOG(gMediaPipelineLog, LogLevel::Error, ("AppendData failed"));
+        // we can't un-read the data, but that's ok since we don't want to
+        // buffer - but don't i-loop!
+        break;
+      }
+    }
+  }
+
+  const RefPtr<MediaSessionConduit> mConduit;
+  // This conduit's sampling rate. This is either 16, 32, 44.1 or 48kHz, and
+  // tries to be the same as the graph rate. If the graph rate is higher than
+  // 48kHz, mRate is capped to 48kHz. If mRate does not match the graph rate,
+  // audio is resampled to the graph rate.
+  const TrackRate mRate;
+  const RefPtr<TaskQueue> mTaskQueue;
+  // Number of frames of data that has been added to the SourceMediaTrack in
+  // the graph's rate. Graph thread only.
+  TrackTicks mPlayedTicks;
+  // Allocation of an audio frame used as a scratch buffer when reading data out
+  // of libwebrtc for forwarding into the graph. Graph thread only.
+  std::unique_ptr<webrtc::AudioFrame> mAudioFrame;
+  // Principal handle used when appending data to the SourceMediaTrack. Graph
+  // thread only.
+  PrincipalHandle mPrincipalHandle;
+  // Privacy of mPrincipalHandle. Graph thread only.
+  PrincipalPrivacy mPrivacy;
+  // Set to true on the sts thread if privacy is requested when ALPN was
+  // negotiated. Set to false again when mPrincipalHandle is private.
+  Atomic<bool> mForceSilence;
+};
+
+MediaPipelineReceiveAudio::MediaPipelineReceiveAudio(
+    const std::string& aPc, RefPtr<MediaTransportHandler> aTransportHandler,
+    RefPtr<AbstractThread> aCallThread, RefPtr<nsISerialEventTarget> aStsThread,
+    RefPtr<AudioSessionConduit> aConduit, RefPtr<SourceMediaTrack> aSource,
+    TrackingId aTrackingId, PrincipalHandle aPrincipalHandle,
+    PrincipalPrivacy aPrivacy)
+    : MediaPipelineReceive(aPc, std::move(aTransportHandler),
+                           std::move(aCallThread), std::move(aStsThread),
+                           std::move(aConduit)),
+      mListener(aSource ? new PipelineListener(
+                              std::move(aSource), std::move(aTrackingId),
+                              mConduit, std::move(aPrincipalHandle), aPrivacy)
+                        : nullptr) {
+  mDescription = mPc + "| Receive audio";
+  if (mListener) {
+    mListener->Init();
+  }
+}
+
+void MediaPipelineReceiveAudio::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MediaPipelineReceive::Shutdown();
+  if (mListener) {
+    mListener->Shutdown();
+  }
+}
+
+void MediaPipelineReceiveAudio::OnPrivacyRequested_s() {
+  ASSERT_ON_THREAD(mStsThread);
+  if (mListener) {
+    mListener->OnPrivacyRequested_s();
+  }
+}
+
+void MediaPipelineReceiveAudio::SetPrivatePrincipal(PrincipalHandle aHandle) {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mListener) {
+    mListener->SetPrivatePrincipal(std::move(aHandle));
+  }
+}
+
+void MediaPipelineReceiveAudio::UpdateListener() {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mListener) {
+    mListener->SetEnabled(mActive.Ref());
+  }
+}
+
+class MediaPipelineReceiveVideo::PipelineListener
+    : public GenericReceiveListener {
+ public:
+  PipelineListener(RefPtr<SourceMediaTrack> aSource, TrackingId aTrackingId,
+                   PrincipalHandle aPrincipalHandle, PrincipalPrivacy aPrivacy)
+      : GenericReceiveListener(std::move(aSource), std::move(aTrackingId)),
+        mImageContainer(
+            MakeAndAddRef<ImageContainer>(ImageContainer::ASYNCHRONOUS)),
+        mMutex("MediaPipelineReceiveVideo::PipelineListener::mMutex"),
+        mPrincipalHandle(std::move(aPrincipalHandle)),
+        mPrivacy(aPrivacy) {}
+  void OnPrivacyRequested_s() {
+    MutexAutoLock lock(mMutex);
+    if (mPrivacy == PrincipalPrivacy::Private) {
+      return;
+    }
+    mForceDropFrames = true;
+  }
+
+  void SetPrivatePrincipal(PrincipalHandle aHandle) {
+    MutexAutoLock lock(mMutex);
+    if (mPrivacy == PrincipalPrivacy::Private) {
+      return;
+    }
+    mPrincipalHandle = std::move(aHandle);
+    mPrivacy = PrincipalPrivacy::Private;
+    mForceDropFrames = false;
+  }
+
+  void RenderVideoFrame(const webrtc::VideoFrameBuffer& aBuffer,
+                        uint32_t aTimeStamp, int64_t aRenderTime) {
+    PrincipalHandle principal;
+    {
+      MutexAutoLock lock(mMutex);
+      if (mForceDropFrames) {
+        return;
+      }
+      principal = mPrincipalHandle;
+    }
+    RefPtr<Image> image;
+    if (aBuffer.type() == webrtc::VideoFrameBuffer::Type::kNative) {
+      // We assume that only native handles are used with the
+      // WebrtcMediaDataCodec decoder.
+      const ImageBuffer* imageBuffer =
+          static_cast<const ImageBuffer*>(&aBuffer);
+      image = imageBuffer->GetNativeImage();
+    } else {
+      MOZ_ASSERT(aBuffer.type() == webrtc::VideoFrameBuffer::Type::kI420);
+      rtc::scoped_refptr<const webrtc::I420BufferInterface> i420(
+          aBuffer.GetI420());
+
+      MOZ_ASSERT(i420->DataY());
+      // Create a video frame using |buffer|.
+      PerformanceRecorder<CopyVideoStage> rec(
+          "MediaPipelineReceiveVideo::CopyToImage"_ns, mTrackingId,
+          i420->width(), i420->height());
+
+      RefPtr<PlanarYCbCrImage> yuvImage =
+          mImageContainer->CreatePlanarYCbCrImage();
+
+      PlanarYCbCrData yuvData;
+      yuvData.mYChannel = const_cast<uint8_t*>(i420->DataY());
+      yuvData.mYStride = i420->StrideY();
+      MOZ_ASSERT(i420->StrideU() == i420->StrideV());
+      yuvData.mCbCrStride = i420->StrideU();
+      yuvData.mCbChannel = const_cast<uint8_t*>(i420->DataU());
+      yuvData.mCrChannel = const_cast<uint8_t*>(i420->DataV());
+      yuvData.mPictureRect = IntRect(0, 0, i420->width(), i420->height());
+      yuvData.mStereoMode = StereoMode::MONO;
+      // This isn't the best default.
+      yuvData.mYUVColorSpace = gfx::YUVColorSpace::BT601;
+      yuvData.mChromaSubsampling =
+          gfx::ChromaSubsampling::HALF_WIDTH_AND_HEIGHT;
+
+      if (!yuvImage->CopyData(yuvData)) {
+        MOZ_ASSERT(false);
+        return;
+      }
+      rec.Record();
+
+      image = std::move(yuvImage);
+    }
+
+    VideoSegment segment;
+    auto size = image->GetSize();
+    segment.AppendFrame(image.forget(), size, principal);
+    mSource->AppendData(&segment);
+  }
+
+ private:
+  RefPtr<layers::ImageContainer> mImageContainer;
+  Mutex mMutex;
+  PrincipalHandle mPrincipalHandle MOZ_GUARDED_BY(mMutex);
+  PrincipalPrivacy mPrivacy MOZ_GUARDED_BY(mMutex);
+  // Set to true on the sts thread if privacy is requested when ALPN was
+  // negotiated. Set to false again when mPrincipalHandle is private.
+  bool mForceDropFrames MOZ_GUARDED_BY(mMutex) = false;
+};
+
+class MediaPipelineReceiveVideo::PipelineRenderer
+    : public mozilla::VideoRenderer {
+ public:
+  explicit PipelineRenderer(MediaPipelineReceiveVideo* aPipeline)
+      : mPipeline(aPipeline) {}
+
+  void Detach() { mPipeline = nullptr; }
+
+  // Implement VideoRenderer
+  void FrameSizeChange(unsigned int aWidth, unsigned int aHeight) override {}
+  void RenderVideoFrame(const webrtc::VideoFrameBuffer& aBuffer,
+                        uint32_t aTimeStamp, int64_t aRenderTime) override {
+    mPipeline->mListener->RenderVideoFrame(aBuffer, aTimeStamp, aRenderTime);
+  }
+
+ private:
+  MediaPipelineReceiveVideo* mPipeline;  // Raw pointer to avoid cycles
+};
+
+MediaPipelineReceiveVideo::MediaPipelineReceiveVideo(
+    const std::string& aPc, RefPtr<MediaTransportHandler> aTransportHandler,
+    RefPtr<AbstractThread> aCallThread, RefPtr<nsISerialEventTarget> aStsThread,
+    RefPtr<VideoSessionConduit> aConduit, RefPtr<SourceMediaTrack> aSource,
+    TrackingId aTrackingId, PrincipalHandle aPrincipalHandle,
+    PrincipalPrivacy aPrivacy)
+    : MediaPipelineReceive(aPc, std::move(aTransportHandler),
+                           std::move(aCallThread), std::move(aStsThread),
+                           std::move(aConduit)),
+      mRenderer(new PipelineRenderer(this)),
+      mListener(aSource ? new PipelineListener(
+                              std::move(aSource), std::move(aTrackingId),
+                              std::move(aPrincipalHandle), aPrivacy)
+                        : nullptr) {
+  mDescription = mPc + "| Receive video";
+  if (mListener) {
+    mListener->Init();
+  }
+  static_cast<VideoSessionConduit*>(mConduit.get())->AttachRenderer(mRenderer);
+}
+
+void MediaPipelineReceiveVideo::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MediaPipelineReceive::Shutdown();
+  if (mListener) {
+    mListener->Shutdown();
+  }
+
+  // stop generating video and thus stop invoking the PipelineRenderer
+  // and PipelineListener - the renderer has a raw ptr to the Pipeline to
+  // avoid cycles, and the render callbacks are invoked from a different
+  // thread so simple null-checks would cause TSAN bugs without locks.
+  static_cast<VideoSessionConduit*>(mConduit.get())->DetachRenderer();
+}
+
+void MediaPipelineReceiveVideo::OnPrivacyRequested_s() {
+  ASSERT_ON_THREAD(mStsThread);
+  if (mListener) {
+    mListener->OnPrivacyRequested_s();
+  }
+}
+
+void MediaPipelineReceiveVideo::SetPrivatePrincipal(PrincipalHandle aHandle) {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mListener) {
+    mListener->SetPrivatePrincipal(std::move(aHandle));
+  }
+}
+
+void MediaPipelineReceiveVideo::UpdateListener() {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mListener) {
+    mListener->SetEnabled(mActive.Ref());
+  }
+}
+
+const dom::RTCStatsTimestampMaker& MediaPipeline::GetTimestampMaker() const {
+  return mConduit->GetTimestampMaker();
+}
+
+DOMHighResTimeStamp MediaPipeline::RtpCSRCStats::GetExpiryFromTime(
+    const DOMHighResTimeStamp aTime) {
+  // DOMHighResTimeStamp is a unit measured in ms
+  return aTime + EXPIRY_TIME_MILLISECONDS;
+}
+
+MediaPipeline::RtpCSRCStats::RtpCSRCStats(const uint32_t aCsrc,
+                                          const DOMHighResTimeStamp aTime)
+    : mCsrc(aCsrc), mTimestamp(aTime) {}
+
+void MediaPipeline::RtpCSRCStats::GetWebidlInstance(
+    dom::RTCRTPContributingSourceStats& aWebidlObj,
+    const nsString& aInboundRtpStreamId) const {
+  nsString statId = u"csrc_"_ns + aInboundRtpStreamId;
+  statId.AppendLiteral("_");
+  statId.AppendInt(mCsrc);
+  aWebidlObj.mId.Construct(statId);
+  aWebidlObj.mType.Construct(RTCStatsType::Csrc);
+  aWebidlObj.mTimestamp.Construct(mTimestamp);
+  aWebidlObj.mContributorSsrc.Construct(mCsrc);
+  aWebidlObj.mInboundRtpStreamId.Construct(aInboundRtpStreamId);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transportbridge/MediaPipeline.h b/dom/media/webrtc/transportbridge/MediaPipeline.h
new file mode 100644
index 0000000000..d58fd12ea3
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/MediaPipeline.h
@@ -0,0 +1,454 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* 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/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef mediapipeline_h__
+#define mediapipeline_h__
+
+#include <map>
+
+#include "transport/sigslot.h"
+#include "transport/transportlayer.h"  // For TransportLayer::State
+
+#include "libwebrtcglue/MediaConduitControl.h"
+#include "mozilla/ReentrantMonitor.h"
+#include "mozilla/Atomics.h"
+#include "mozilla/StateMirroring.h"
+#include "transport/mediapacket.h"
+#include "transport/runnable_utils.h"
+#include "AudioPacketizer.h"
+#include "MediaEventSource.h"
+#include "MediaPipelineFilter.h"
+#include "MediaSegment.h"
+#include "PrincipalChangeObserver.h"
+#include "jsapi/PacketDumper.h"
+#include "PerformanceRecorder.h"
+
+// Should come from MediaEngine.h, but that's a pain to include here
+// because of the MOZILLA_EXTERNAL_LINKAGE stuff.
+#define WEBRTC_MAX_SAMPLE_RATE 48000
+
+class nsIPrincipal;
+
+namespace webrtc {
+struct RTPHeader;
+class RtpHeaderExtensionMap;
+class RtpPacketReceived;
+}  // namespace webrtc
+
+namespace mozilla {
+class AudioProxyThread;
+class MediaInputPort;
+class MediaPipelineFilter;
+class MediaTransportHandler;
+class PeerIdentity;
+class ProcessedMediaTrack;
+class SourceMediaTrack;
+class VideoFrameConverter;
+class MediaSessionConduit;
+class AudioSessionConduit;
+class VideoSessionConduit;
+
+namespace dom {
+class MediaStreamTrack;
+struct RTCRTPContributingSourceStats;
+class RTCStatsTimestampMaker;
+}  // namespace dom
+
+struct MediaPipelineReceiveControlInterface {
+  virtual AbstractCanonical<bool>* CanonicalReceiving() = 0;
+};
+
+struct MediaPipelineTransmitControlInterface {
+  virtual AbstractCanonical<bool>* CanonicalTransmitting() = 0;
+};
+
+// A class that represents the pipeline of audio and video
+// The dataflow looks like:
+//
+// TRANSMIT
+// CaptureDevice -> stream -> [us] -> conduit -> [us] -> transport -> network
+//
+// RECEIVE
+// network -> transport -> [us] -> conduit -> [us] -> stream -> Playout
+//
+// The boxes labeled [us] are just bridge logic implemented in this class
+//
+// We have to deal with a number of threads:
+//
+// GSM:
+//   * Assembles the pipeline
+// SocketTransportService
+//   * Receives notification that ICE and DTLS have completed
+//   * Processes incoming network data and passes it to the conduit
+//   * Processes outgoing RTP and RTCP
+// MediaTrackGraph
+//   * Receives outgoing data from the MediaTrackGraph
+//   * Receives pull requests for more data from the
+//     MediaTrackGraph
+// One or another GIPS threads
+//   * Receives RTCP messages to send to the other side
+//   * Processes video frames GIPS wants to render
+//
+// For a transmitting conduit, "output" is RTP and "input" is RTCP.
+// For a receiving conduit, "input" is RTP and "output" is RTCP.
+//
+
+class MediaPipeline : public sigslot::has_slots<> {
+ public:
+  enum class DirectionType { TRANSMIT, RECEIVE };
+  MediaPipeline(const std::string& aPc,
+                RefPtr<MediaTransportHandler> aTransportHandler,
+                DirectionType aDirection, RefPtr<AbstractThread> aCallThread,
+                RefPtr<nsISerialEventTarget> aStsThread,
+                RefPtr<MediaSessionConduit> aConduit);
+
+  void SetLevel(size_t aLevel) { mLevel = aLevel; }
+
+  // Main thread shutdown.
+  virtual void Shutdown();
+
+  void UpdateTransport_m(const std::string& aTransportId,
+                         UniquePtr<MediaPipelineFilter>&& aFilter);
+
+  void UpdateTransport_s(const std::string& aTransportId,
+                         UniquePtr<MediaPipelineFilter>&& aFilter);
+
+  virtual DirectionType Direction() const { return mDirection; }
+  size_t Level() const { return mLevel; }
+  virtual bool IsVideo() const = 0;
+
+  class RtpCSRCStats {
+   public:
+    // Gets an expiration time for CRC info given a reference time,
+    //   this reference time would normally be the time of calling.
+    //   This value can then be used to check if a RtpCSRCStats
+    //   has expired via Expired(...)
+    static DOMHighResTimeStamp GetExpiryFromTime(
+        const DOMHighResTimeStamp aTime);
+
+    RtpCSRCStats(const uint32_t aCsrc, const DOMHighResTimeStamp aTime);
+    ~RtpCSRCStats() = default;
+    // Initialize a webidl representation suitable for adding to a report.
+    //   This assumes that the webidl object is empty.
+    // @param aWebidlObj the webidl binding object to popluate
+    // @param aInboundRtpStreamId the associated RTCInboundRTPStreamStats.id
+    void GetWebidlInstance(dom::RTCRTPContributingSourceStats& aWebidlObj,
+                           const nsString& aInboundRtpStreamId) const;
+    void SetTimestamp(const DOMHighResTimeStamp aTime) { mTimestamp = aTime; }
+    // Check if the RtpCSRCStats has expired, checks against a
+    //   given expiration time.
+    bool Expired(const DOMHighResTimeStamp aExpiry) const {
+      return mTimestamp < aExpiry;
+    }
+
+   private:
+    static const double constexpr EXPIRY_TIME_MILLISECONDS = 10 * 1000;
+    const uint32_t mCsrc;
+    DOMHighResTimeStamp mTimestamp;
+  };
+
+  // Gets the gathered contributing source stats for the last expiration period.
+  // @param aId the stream id to use for populating inboundRtpStreamId field
+  // @param aArr the array to append the stats objects to
+  void GetContributingSourceStats(
+      const nsString& aInboundRtpStreamId,
+      FallibleTArray<dom::RTCRTPContributingSourceStats>& aArr) const;
+
+  int32_t RtpPacketsSent() const { return mRtpPacketsSent; }
+  int64_t RtpBytesSent() const { return mRtpBytesSent; }
+  int32_t RtcpPacketsSent() const { return mRtcpPacketsSent; }
+  int32_t RtpPacketsReceived() const { return mRtpPacketsReceived; }
+  int64_t RtpBytesReceived() const { return mRtpBytesReceived; }
+  int32_t RtcpPacketsReceived() const { return mRtcpPacketsReceived; }
+
+  const dom::RTCStatsTimestampMaker& GetTimestampMaker() const;
+
+  // Thread counting
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaPipeline)
+
+ protected:
+  virtual ~MediaPipeline();
+
+  // The transport is ready
+  virtual void TransportReady_s() {}
+
+  void IncrementRtpPacketsSent(const MediaPacket& aPacket);
+  void IncrementRtcpPacketsSent();
+  void IncrementRtpPacketsReceived(int aBytes);
+  void IncrementRtcpPacketsReceived();
+
+  virtual void SendPacket(MediaPacket&& packet);
+
+  // Process slots on transports
+  void RtpStateChange(const std::string& aTransportId, TransportLayer::State);
+  void RtcpStateChange(const std::string& aTransportId, TransportLayer::State);
+  virtual void CheckTransportStates();
+  void PacketReceived(const std::string& aTransportId,
+                      const MediaPacket& packet);
+  void AlpnNegotiated(const std::string& aAlpn, bool aPrivacyRequested);
+
+  void RtpPacketReceived(const MediaPacket& packet);
+  void RtcpPacketReceived(const MediaPacket& packet);
+
+  void EncryptedPacketSending(const std::string& aTransportId,
+                              const MediaPacket& aPacket);
+
+  void SetDescription_s(const std::string& description);
+
+ public:
+  const RefPtr<MediaSessionConduit> mConduit;
+  const DirectionType mDirection;
+
+  // Pointers to the threads we need. Initialized at creation
+  // and used all over the place.
+  const RefPtr<AbstractThread> mCallThread;
+  const RefPtr<nsISerialEventTarget> mStsThread;
+
+ protected:
+  // True if we should be actively transmitting or receiving data. Main thread
+  // only.
+  Mirror<bool> mActive;
+  Atomic<size_t> mLevel;
+  std::string mTransportId;
+  const RefPtr<MediaTransportHandler> mTransportHandler;
+
+  TransportLayer::State mRtpState = TransportLayer::TS_NONE;
+  TransportLayer::State mRtcpState = TransportLayer::TS_NONE;
+  bool mSignalsConnected = false;
+
+  // Only safe to access from STS thread.
+  int32_t mRtpPacketsSent;
+  int32_t mRtcpPacketsSent;
+  int32_t mRtpPacketsReceived;
+  int32_t mRtcpPacketsReceived;
+  int64_t mRtpBytesSent;
+  int64_t mRtpBytesReceived;
+
+  // Only safe to access from STS thread.
+  std::map<uint32_t, RtpCSRCStats> mCsrcStats;
+
+  // Written in c'tor. Read on STS and main thread.
+  const std::string mPc;
+
+  // String describing this MediaPipeline for logging purposes. Only safe to
+  // access from STS thread.
+  std::string mDescription;
+
+  // Written in c'tor, all following accesses are on the STS thread.
+  UniquePtr<MediaPipelineFilter> mFilter;
+  const UniquePtr<webrtc::RtpHeaderExtensionMap> mRtpHeaderExtensionMap;
+
+  RefPtr<PacketDumper> mPacketDumper;
+
+  MediaEventProducerExc<webrtc::RtpPacketReceived, webrtc::RTPHeader>
+      mRtpReceiveEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtcpReceiveEvent;
+  MediaEventProducerExc<MediaPacket> mReceiverRtcpReceiveEvent;
+
+  MediaEventListener mRtpSendEventListener;
+  MediaEventListener mSenderRtcpSendEventListener;
+  MediaEventListener mReceiverRtcpSendEventListener;
+
+ private:
+  bool IsRtp(const unsigned char* aData, size_t aLen) const;
+  // Must be called on the STS thread.  Must be called after Shutdown().
+  void DetachTransport_s();
+};
+
+// A specialization of pipeline for reading from an input device
+// and transmitting to the network.
+class MediaPipelineTransmit
+    : public MediaPipeline,
+      public dom::PrincipalChangeObserver<dom::MediaStreamTrack> {
+ public:
+  // Set aRtcpTransport to nullptr to use rtcp-mux
+  MediaPipelineTransmit(const std::string& aPc,
+                        RefPtr<MediaTransportHandler> aTransportHandler,
+                        RefPtr<AbstractThread> aCallThread,
+                        RefPtr<nsISerialEventTarget> aStsThread, bool aIsVideo,
+                        RefPtr<MediaSessionConduit> aConduit);
+
+  void InitControl(MediaPipelineTransmitControlInterface* aControl);
+
+  void Shutdown() override;
+
+  bool Transmitting() const;
+
+  // written and used from MainThread
+  bool IsVideo() const override;
+
+  // When the principal of the domtrack changes, it calls through to here
+  // so that we can determine whether to enable track transmission.
+  // In cases where the peer isn't yet identified, we disable the pipeline (not
+  // the stream, that would potentially affect others), so that it sends
+  // black/silence.  Once the peer is identified, re-enable those streams.
+  virtual void UpdateSinkIdentity(nsIPrincipal* aPrincipal,
+                                  const PeerIdentity* aSinkIdentity);
+
+  // for monitoring changes in track ownership
+  void PrincipalChanged(dom::MediaStreamTrack* aTrack) override;
+
+  // Override MediaPipeline::TransportReady_s.
+  void TransportReady_s() override;
+
+  // Replace a track with a different one.
+  nsresult SetTrack(const RefPtr<dom::MediaStreamTrack>& aDomTrack);
+
+  // Used to correlate stats
+  RefPtr<dom::MediaStreamTrack> GetTrack() const;
+
+  // For test use only. This allows a send track to be set without a
+  // corresponding dom track.
+  void SetSendTrackOverride(const RefPtr<ProcessedMediaTrack>& aSendTrack);
+
+  // Separate classes to allow ref counting
+  class PipelineListener;
+  class VideoFrameFeeder;
+
+ protected:
+  ~MediaPipelineTransmit();
+
+  // Updates mDescription (async) with information about the track we are
+  // transmitting.
+  std::string GenerateDescription() const;
+
+  // Sets up mSendPort and mSendTrack to feed mConduit if we are transmitting
+  // and have a dom track but no send track. Main thread only.
+  void UpdateSendState();
+
+ private:
+  WatchManager<MediaPipelineTransmit> mWatchManager;
+  const bool mIsVideo;
+  const RefPtr<PipelineListener> mListener;
+  RefPtr<AudioProxyThread> mAudioProcessing;
+  RefPtr<VideoFrameConverter> mConverter;
+  MediaEventListener mFrameListener;
+  Watchable<RefPtr<dom::MediaStreamTrack>> mDomTrack;
+  // Input port connecting mDomTrack's MediaTrack to mSendTrack.
+  RefPtr<MediaInputPort> mSendPort;
+  // The source track of the mSendTrack. Main thread only.
+  RefPtr<ProcessedMediaTrack> mSendPortSource;
+  // True if a parameter affecting mDescription has changed. To avoid updating
+  // the description unnecessarily. Main thread only.
+  bool mDescriptionInvalidated = true;
+  // Set true once we trigger the async removal of mSendTrack. Set false once
+  // the async removal is done. Main thread only.
+  bool mUnsettingSendTrack = false;
+  // MediaTrack that we send over the network. This allows changing mDomTrack.
+  // Because changing mSendTrack is async and can be racy (when changing from a
+  // track in one graph to a track in another graph), it is set very strictly.
+  // If mSendTrack is null it can be set by UpdateSendState().
+  // If it is non-null it can only be set to null, and only by the
+  // RemoveListener MozPromise handler, as seen in UpdateSendState.
+  RefPtr<ProcessedMediaTrack> mSendTrack;
+  // When this is set and we are active, this track will be used as mSendTrack.
+  // Allows unittests to insert a send track without requiring a dom track or a
+  // graph. Main thread only.
+  Watchable<RefPtr<ProcessedMediaTrack>> mSendTrackOverride;
+  // True when mSendTrack is set, not destroyed and mActive is true. mListener
+  // is attached to mSendTrack when this is true. Main thread only.
+  bool mTransmitting = false;
+};
+
+// A specialization of pipeline for reading from the network and
+// rendering media.
+class MediaPipelineReceive : public MediaPipeline {
+ public:
+  // Set aRtcpTransport to nullptr to use rtcp-mux
+  MediaPipelineReceive(const std::string& aPc,
+                       RefPtr<MediaTransportHandler> aTransportHandler,
+                       RefPtr<AbstractThread> aCallThread,
+                       RefPtr<nsISerialEventTarget> aStsThread,
+                       RefPtr<MediaSessionConduit> aConduit);
+
+  void InitControl(MediaPipelineReceiveControlInterface* aControl);
+
+  // Called when ALPN is negotiated and is requesting privacy, so receive
+  // pipelines do not enter data into the graph under a content principal.
+  virtual void OnPrivacyRequested_s() = 0;
+
+  // Called after privacy has been requested, with the updated private
+  // principal.
+  virtual void SetPrivatePrincipal(PrincipalHandle aHandle) = 0;
+
+  void Shutdown() override;
+
+ protected:
+  ~MediaPipelineReceive();
+
+  virtual void UpdateListener() = 0;
+
+ private:
+  WatchManager<MediaPipelineReceive> mWatchManager;
+};
+
+// A specialization of pipeline for reading from the network and
+// rendering audio.
+class MediaPipelineReceiveAudio : public MediaPipelineReceive {
+ public:
+  MediaPipelineReceiveAudio(const std::string& aPc,
+                            RefPtr<MediaTransportHandler> aTransportHandler,
+                            RefPtr<AbstractThread> aCallThread,
+                            RefPtr<nsISerialEventTarget> aStsThread,
+                            RefPtr<AudioSessionConduit> aConduit,
+                            RefPtr<SourceMediaTrack> aSource,
+                            TrackingId aTrackingId,
+                            PrincipalHandle aPrincipalHandle,
+                            PrincipalPrivacy aPrivacy);
+
+  void Shutdown() override;
+
+  bool IsVideo() const override { return false; }
+
+  void OnPrivacyRequested_s() override;
+  void SetPrivatePrincipal(PrincipalHandle aHandle) override;
+
+ private:
+  void UpdateListener() override;
+
+  // Separate class to allow ref counting
+  class PipelineListener;
+
+  const RefPtr<PipelineListener> mListener;
+};
+
+// A specialization of pipeline for reading from the network and
+// rendering video.
+class MediaPipelineReceiveVideo : public MediaPipelineReceive {
+ public:
+  MediaPipelineReceiveVideo(const std::string& aPc,
+                            RefPtr<MediaTransportHandler> aTransportHandler,
+                            RefPtr<AbstractThread> aCallThread,
+                            RefPtr<nsISerialEventTarget> aStsThread,
+                            RefPtr<VideoSessionConduit> aConduit,
+                            RefPtr<SourceMediaTrack> aSource,
+                            TrackingId aTrackingId,
+                            PrincipalHandle aPrincipalHandle,
+                            PrincipalPrivacy aPrivacy);
+
+  void Shutdown() override;
+
+  bool IsVideo() const override { return true; }
+
+  void OnPrivacyRequested_s() override;
+  void SetPrivatePrincipal(PrincipalHandle aHandle) override;
+
+ private:
+  void UpdateListener() override;
+
+  class PipelineRenderer;
+  friend class PipelineRenderer;
+
+  // Separate class to allow ref counting
+  class PipelineListener;
+
+  const RefPtr<PipelineRenderer> mRenderer;
+  const RefPtr<PipelineListener> mListener;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp b/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp
new file mode 100644
index 0000000000..1acb73e9f2
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp
@@ -0,0 +1,153 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: softtabstop=2:shiftwidth=2:expandtab
+ * */
+/* 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/. */
+
+// Original author: bcampen@mozilla.com
+
+#include "MediaPipelineFilter.h"
+
+#include "api/rtp_headers.h"
+#include "api/rtp_parameters.h"
+#include "mozilla/Logging.h"
+
+// defined in MediaPipeline.cpp
+extern mozilla::LazyLogModule gMediaPipelineLog;
+
+#define DEBUG_LOG(x) MOZ_LOG(gMediaPipelineLog, LogLevel::Debug, x)
+
+namespace mozilla {
+MediaPipelineFilter::MediaPipelineFilter(
+    const std::vector<webrtc::RtpExtension>& aExtMap)
+    : mExtMap(aExtMap) {}
+
+void MediaPipelineFilter::SetRemoteMediaStreamId(
+    const Maybe<std::string>& aMid) {
+  if (aMid != mRemoteMid) {
+    DEBUG_LOG(("MediaPipelineFilter added new remote RTP MID: '%s'.",
+               aMid.valueOr("").c_str()));
+    mRemoteMid = aMid;
+    mRemoteMidBindings.clear();
+  }
+}
+
+bool MediaPipelineFilter::Filter(const webrtc::RTPHeader& header) {
+  DEBUG_LOG(("MediaPipelineFilter inspecting seq# %u SSRC: %u",
+             header.sequenceNumber, header.ssrc));
+
+  auto fromStreamId = [](const std::string& aId) {
+    return Maybe<std::string>(aId.empty() ? Nothing() : Some(aId));
+  };
+
+  //
+  //  MID Based Filtering
+  //
+
+  const auto mid = fromStreamId(header.extension.mid);
+
+  // Check to see if a bound SSRC is moved to a new MID
+  if (mRemoteMidBindings.count(header.ssrc) == 1 && mid && mRemoteMid != mid) {
+    mRemoteMidBindings.erase(header.ssrc);
+  }
+  // Bind an SSRC if a matching MID is found
+  if (mid && mRemoteMid == mid) {
+    DEBUG_LOG(("MediaPipelineFilter learned SSRC: %u for MID: '%s'",
+               header.ssrc, mRemoteMid.value().c_str()));
+    mRemoteMidBindings.insert(header.ssrc);
+  }
+  // Check for matching MID
+  if (!mRemoteMidBindings.empty()) {
+    MOZ_ASSERT(mRemoteMid != Nothing());
+    if (mRemoteMidBindings.count(header.ssrc) == 1) {
+      DEBUG_LOG(
+          ("MediaPipelineFilter SSRC: %u matched for MID: '%s'."
+           " passing packet",
+           header.ssrc, mRemoteMid.value().c_str()));
+      return true;
+    }
+    DEBUG_LOG(
+        ("MediaPipelineFilter SSRC: %u did not match bound SSRC(s) for"
+         " MID: '%s'. ignoring packet",
+         header.ssrc, mRemoteMid.value().c_str()));
+    for (const uint32_t ssrc : mRemoteMidBindings) {
+      DEBUG_LOG(("MID %s is associated with SSRC: %u",
+                 mRemoteMid.value().c_str(), ssrc));
+    }
+    return false;
+  }
+
+  //
+  // RTP-STREAM-ID based filtering (for tests only)
+  //
+
+  //
+  // Remote SSRC based filtering
+  //
+
+  if (remote_ssrc_set_.count(header.ssrc)) {
+    DEBUG_LOG(
+        ("MediaPipelineFilter SSRC: %u matched remote SSRC set."
+         " passing packet",
+         header.ssrc));
+    return true;
+  }
+  DEBUG_LOG(
+      ("MediaPipelineFilter SSRC: %u did not match any of %zu"
+       " remote SSRCS.",
+       header.ssrc, remote_ssrc_set_.size()));
+
+  //
+  // PT, payload type, last ditch effort filtering
+  //
+
+  if (payload_type_set_.count(header.payloadType)) {
+    DEBUG_LOG(
+        ("MediaPipelineFilter payload-type: %u matched %zu"
+         " unique payload type. learning ssrc. passing packet",
+         header.ssrc, remote_ssrc_set_.size()));
+    // Actual match. We need to update the ssrc map so we can route rtcp
+    // sender reports correctly (these use a different payload-type field)
+    AddRemoteSSRC(header.ssrc);
+    return true;
+  }
+  DEBUG_LOG(
+      ("MediaPipelineFilter payload-type: %u did not match any of %zu"
+       " unique payload-types.",
+       header.payloadType, payload_type_set_.size()));
+  DEBUG_LOG(
+      ("MediaPipelineFilter packet failed to match any criteria."
+       " ignoring packet"));
+  return false;
+}
+
+void MediaPipelineFilter::AddRemoteSSRC(uint32_t ssrc) {
+  remote_ssrc_set_.insert(ssrc);
+}
+
+void MediaPipelineFilter::AddUniquePT(uint8_t payload_type) {
+  payload_type_set_.insert(payload_type);
+}
+
+void MediaPipelineFilter::Update(const MediaPipelineFilter& filter_update) {
+  // We will not stomp the remote_ssrc_set_ if the update has no ssrcs,
+  // because we don't want to unlearn any remote ssrcs unless the other end
+  // has explicitly given us a new set.
+  if (!filter_update.remote_ssrc_set_.empty()) {
+    remote_ssrc_set_ = filter_update.remote_ssrc_set_;
+  }
+  // We don't want to overwrite the learned binding unless we have changed MIDs
+  // or the update contains a MID binding.
+  if (!filter_update.mRemoteMidBindings.empty() ||
+      (filter_update.mRemoteMid && filter_update.mRemoteMid != mRemoteMid)) {
+    mRemoteMid = filter_update.mRemoteMid;
+    mRemoteMidBindings = filter_update.mRemoteMidBindings;
+  }
+  payload_type_set_ = filter_update.payload_type_set_;
+
+  // Use extmapping from new filter
+  mExtMap = filter_update.mExtMap;
+}
+
+}  // end namespace mozilla
diff --git a/dom/media/webrtc/transportbridge/MediaPipelineFilter.h b/dom/media/webrtc/transportbridge/MediaPipelineFilter.h
new file mode 100644
index 0000000000..9b40bceda8
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/MediaPipelineFilter.h
@@ -0,0 +1,89 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: softtabstop=2:shiftwidth=2:expandtab
+ * */
+/* 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/. */
+
+// Original author: bcampen@mozilla.com
+
+#ifndef mediapipelinefilter_h__
+#define mediapipelinefilter_h__
+
+#include <cstddef>
+#include <stdint.h>
+#include <string>
+
+#include <set>
+#include <vector>
+
+#include "mozilla/Maybe.h"
+
+namespace webrtc {
+struct RTPHeader;
+struct RtpExtension;
+}  // namespace webrtc
+
+namespace mozilla {
+
+// TODO @@NG update documentation after initial review
+
+// A class that handles the work of filtering RTP packets that arrive at a
+// MediaPipeline. This is primarily important for the use of BUNDLE (ie;
+// multiple m-lines share the same RTP stream). There are three ways that this
+// can work;
+//
+// 1) In our SDP, we include a media-level extmap parameter with a unique
+//    integer of our choosing, with the hope that the other side will include
+//    this value in a header in the first few RTP packets it sends us. This
+//    allows us to perform correlation in cases where the other side has not
+//    informed us of the ssrcs it will be sending (either because it did not
+//    include them in its SDP, or their SDP has not arrived yet)
+//    and also gives us the opportunity to learn SSRCs from packets so adorned.
+//
+// 2) If the remote endpoint includes SSRC media-level attributes in its SDP,
+//    we can simply use this information to populate the filter. The only
+//    shortcoming here is when RTP packets arrive before the answer does. See
+//    above.
+//
+// 3) As a fallback, we can try to use payload type IDs to perform correlation,
+//    but only when the type id is unique to this media section.
+//    This too allows us to learn about SSRCs (mostly useful for filtering
+//    sender reports later).
+class MediaPipelineFilter {
+ public:
+  MediaPipelineFilter() = default;
+  explicit MediaPipelineFilter(
+      const std::vector<webrtc::RtpExtension>& aExtMap);
+
+  // Checks whether this packet passes the filter, possibly updating the filter
+  // in the process (if the MID or payload types are used, they can teach
+  // the filter about ssrcs)
+  bool Filter(const webrtc::RTPHeader& header);
+
+  void AddRemoteSSRC(uint32_t ssrc);
+
+  void SetRemoteMediaStreamId(const Maybe<std::string>& aMid);
+
+  // When a payload type id is unique to our media section, add it here.
+  void AddUniquePT(uint8_t payload_type);
+
+  void Update(const MediaPipelineFilter& filter_update);
+
+  std::vector<webrtc::RtpExtension> GetExtmap() const { return mExtMap; }
+
+ private:
+  // The number of filters we manage here is quite small, so I am optimizing
+  // for readability.
+  std::set<uint32_t> remote_ssrc_set_;
+  std::set<uint8_t> payload_type_set_;
+  Maybe<std::string> mRemoteMid;
+  std::set<uint32_t> mRemoteMidBindings;
+  // RID extension can be set by tests and is sticky, the rest of
+  // the mapping is not.
+  std::vector<webrtc::RtpExtension> mExtMap;
+};
+
+}  // end namespace mozilla
+
+#endif  // mediapipelinefilter_h__
diff --git a/dom/media/webrtc/transportbridge/RtpLogger.cpp b/dom/media/webrtc/transportbridge/RtpLogger.cpp
new file mode 100644
index 0000000000..aac1fad197
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/RtpLogger.cpp
@@ -0,0 +1,67 @@
+/* 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/. */
+
+// Original author: nohlmeier@mozilla.com
+
+#include "RtpLogger.h"
+#include "mozilla/Logging.h"
+
+#include <ctime>
+#include <iomanip>
+#include <sstream>
+#ifdef _WIN32
+#  include <time.h>
+#  include <sys/timeb.h>
+#else
+#  include <sys/time.h>
+#endif
+
+// Logging context
+using namespace mozilla;
+
+mozilla::LazyLogModule gRtpLoggerLog("RtpLogger");
+
+namespace mozilla {
+
+bool RtpLogger::IsPacketLoggingOn() {
+  return MOZ_LOG_TEST(gRtpLoggerLog, LogLevel::Debug);
+}
+
+void RtpLogger::LogPacket(const MediaPacket& packet, bool input,
+                          std::string desc) {
+  if (MOZ_LOG_TEST(gRtpLoggerLog, LogLevel::Debug)) {
+    bool isRtp = (packet.type() == MediaPacket::RTP);
+    std::stringstream ss;
+    /* This creates text2pcap compatible format, e.g.:
+     *  RTCP_PACKET O 10:36:26.864934  000000 80 c8 00 06 6d ...
+     */
+    ss << (input ? "I " : "O ");
+    std::time_t t = std::time(nullptr);
+    std::tm tm = *std::localtime(&t);
+    char buf[9];
+    if (0 < strftime(buf, sizeof(buf), "%H:%M:%S", &tm)) {
+      ss << buf;
+    }
+    ss << std::setfill('0');
+#ifdef _WIN32
+    struct timeb tb;
+    ftime(&tb);
+    ss << "." << (tb.millitm) << " ";
+#else
+    struct timeval tv;
+    gettimeofday(&tv, NULL);
+    ss << "." << (tv.tv_usec) << " ";
+#endif
+    ss << " 000000";
+    ss << std::hex << std::setfill('0');
+    for (size_t i = 0; i < packet.len(); ++i) {
+      ss << " " << std::setw(2) << (int)packet.data()[i];
+    }
+    MOZ_LOG(gRtpLoggerLog, LogLevel::Debug,
+            ("%s%s%s", desc.c_str(), (isRtp ? " RTP_PACKET " : " RTCP_PACKET "),
+             ss.str().c_str()));
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transportbridge/RtpLogger.h b/dom/media/webrtc/transportbridge/RtpLogger.h
new file mode 100644
index 0000000000..fcfaede6e2
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/RtpLogger.h
@@ -0,0 +1,28 @@
+/* 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/. */
+
+// Original author: nohlmeier@mozilla.com
+
+#ifndef rtplogger_h__
+#define rtplogger_h__
+
+#include "transport/mediapacket.h"
+
+namespace mozilla {
+
+/* This class logs RTP and RTCP packets in hex in a format compatible to
+ * text2pcap.
+ * Example to convert the MOZ log file into a PCAP file:
+ *   egrep '(RTP_PACKET|RTCP_PACKET)' moz.log | \
+ *     text2pcap -D -n -l 1 -i 17 -u 1234,1235 -t '%H:%M:%S.' - rtp.pcap
+ */
+class RtpLogger {
+ public:
+  static bool IsPacketLoggingOn();
+  static void LogPacket(const MediaPacket& packet, bool input,
+                        std::string desc);
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transportbridge/moz.build b/dom/media/webrtc/transportbridge/moz.build
new file mode 100644
index 0000000000..290dd26a0a
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/moz.build
@@ -0,0 +1,27 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# 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("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "/dom/media",
+    "/dom/media/webrtc",
+    "/ipc/chromium/src",
+    "/media/libyuv/libyuv/include",
+    "/media/webrtc",
+    "/third_party/libsrtp/src/crypto/include",
+    "/third_party/libsrtp/src/include",
+    "/third_party/libwebrtc",
+    "/third_party/libwebrtc/third_party/abseil-cpp",
+]
+
+UNIFIED_SOURCES += [
+    "MediaPipeline.cpp",
+    "MediaPipelineFilter.cpp",
+    "RtpLogger.cpp",
+]
+
+FINAL_LIBRARY = "xul"