Add m-esr52 at 52.6.0

6 files changed, 3358 insertions, 0 deletions

diff --git a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
new file mode 100644
index 000000000..586876406
--- /dev/null
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
@@ -0,0 +1,2377 @@
+/* -*- 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"
+
+#ifndef USE_FAKE_MEDIA_STREAMS
+#include "MediaStreamGraphImpl.h"
+#endif
+
+#include <math.h>
+
+#include "nspr.h"
+#include "srtp.h"
+
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+#include "VideoSegment.h"
+#include "Layers.h"
+#include "LayersLogging.h"
+#include "ImageTypes.h"
+#include "ImageContainer.h"
+#include "DOMMediaStream.h"
+#include "MediaStreamTrack.h"
+#include "MediaStreamListener.h"
+#include "MediaStreamVideoSink.h"
+#include "VideoUtils.h"
+#include "VideoStreamTrack.h"
+#ifdef WEBRTC_GONK
+#include "GrallocImages.h"
+#include "mozilla/layers/GrallocTextureClient.h"
+#endif
+#endif
+
+#include "nsError.h"
+#include "AudioSegment.h"
+#include "MediaSegment.h"
+#include "MediaPipelineFilter.h"
+#include "databuffer.h"
+#include "transportflow.h"
+#include "transportlayer.h"
+#include "transportlayerdtls.h"
+#include "transportlayerice.h"
+#include "runnable_utils.h"
+#include "libyuv/convert.h"
+#include "mozilla/SharedThreadPool.h"
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+#include "mozilla/PeerIdentity.h"
+#include "mozilla/TaskQueue.h"
+#endif
+#include "mozilla/gfx/Point.h"
+#include "mozilla/gfx/Types.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/UniquePtrExtensions.h"
+#include "mozilla/Sprintf.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/common_video/interface/native_handle.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/video_engine/include/vie_errors.h"
+
+#include "logging.h"
+
+// Max size given stereo is 480*2*2 = 1920 (10ms of 16-bits stereo audio at
+// 48KHz)
+#define AUDIO_SAMPLE_BUFFER_MAX 480*2*2
+static_assert((WEBRTC_DEFAULT_SAMPLE_RATE/100)*sizeof(uint16_t) * 2
+               <= AUDIO_SAMPLE_BUFFER_MAX,
+               "AUDIO_SAMPLE_BUFFER_MAX is not large enough");
+
+using namespace mozilla;
+using namespace mozilla::dom;
+using namespace mozilla::gfx;
+using namespace mozilla::layers;
+
+// Logging context
+MOZ_MTLOG_MODULE("mediapipeline")
+
+namespace mozilla {
+
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+class VideoConverterListener
+{
+public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(VideoConverterListener)
+
+  virtual void OnVideoFrameConverted(unsigned char* aVideoFrame,
+                                     unsigned int aVideoFrameLength,
+                                     unsigned short aWidth,
+                                     unsigned short aHeight,
+                                     VideoType aVideoType,
+                                     uint64_t aCaptureTime) = 0;
+
+  virtual void OnVideoFrameConverted(webrtc::I420VideoFrame& aVideoFrame) = 0;
+
+protected:
+  virtual ~VideoConverterListener() {}
+};
+
+// I420 buffer size macros
+#define YSIZE(x,y) ((x)*(y))
+#define CRSIZE(x,y) ((((x)+1) >> 1) * (((y)+1) >> 1))
+#define I420SIZE(x,y) (YSIZE((x),(y)) + 2 * CRSIZE((x),(y)))
+
+// An async video frame format converter.
+//
+// Input is typically a MediaStream(Track)Listener driven by MediaStreamGraph.
+//
+// We keep track of the size of the TaskQueue so we can drop frames if
+// conversion is taking too long.
+//
+// Output is passed through to all added VideoConverterListeners on a TaskQueue
+// thread whenever a frame is converted.
+class VideoFrameConverter
+{
+public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(VideoFrameConverter)
+
+  VideoFrameConverter()
+    : mLength(0)
+    , last_img_(-1) // -1 is not a guaranteed invalid serial. See bug 1262134.
+    , disabled_frame_sent_(false)
+#ifdef DEBUG
+    , mThrottleCount(0)
+    , mThrottleRecord(0)
+#endif
+    , mMutex("VideoFrameConverter")
+  {
+    MOZ_COUNT_CTOR(VideoFrameConverter);
+
+    RefPtr<SharedThreadPool> pool =
+      SharedThreadPool::Get(NS_LITERAL_CSTRING("VideoFrameConverter"));
+
+    mTaskQueue = MakeAndAddRef<TaskQueue>(pool.forget());
+  }
+
+  void QueueVideoChunk(VideoChunk& aChunk, bool aForceBlack)
+  {
+    if (aChunk.IsNull()) {
+      return;
+    }
+
+    // We get passed duplicate frames every ~10ms even with no frame change.
+    int32_t serial = aChunk.mFrame.GetImage()->GetSerial();
+    if (serial == last_img_) {
+      return;
+    }
+    last_img_ = serial;
+
+    // A throttling limit of 1 allows us to convert 2 frames concurrently.
+    // It's short enough to not build up too significant a delay, while
+    // giving us a margin to not cause some machines to drop every other frame.
+    const int32_t queueThrottlingLimit = 1;
+    if (mLength > queueThrottlingLimit) {
+      MOZ_MTLOG(ML_DEBUG, "VideoFrameConverter " << this << " queue is full." <<
+                          " Throttling by throwing away a frame.");
+#ifdef DEBUG
+      ++mThrottleCount;
+      mThrottleRecord = std::max(mThrottleCount, mThrottleRecord);
+#endif
+      return;
+    }
+
+#ifdef DEBUG
+    if (mThrottleCount > 0) {
+      auto level = ML_DEBUG;
+      if (mThrottleCount > 5) {
+        // Log at a higher level when we have large drops.
+        level = ML_INFO;
+      }
+      MOZ_MTLOG(level, "VideoFrameConverter " << this << " stopped" <<
+                       " throttling after throwing away " << mThrottleCount <<
+                       " frames. Longest throttle so far was " <<
+                       mThrottleRecord << " frames.");
+      mThrottleCount = 0;
+    }
+#endif
+
+    bool forceBlack = aForceBlack || aChunk.mFrame.GetForceBlack();
+
+    if (forceBlack) {
+      // Reset the last-img check.
+      // -1 is not a guaranteed invalid serial. See bug 1262134.
+      last_img_ = -1;
+
+      if (disabled_frame_sent_) {
+        // After disabling we just pass one black frame to the encoder.
+        // Allocating and setting it to black steals some performance
+        // that can be avoided. We don't handle resolution changes while
+        // disabled for now.
+        return;
+      }
+
+      disabled_frame_sent_ = true;
+    } else {
+      disabled_frame_sent_ = false;
+    }
+
+    ++mLength; // Atomic
+
+    nsCOMPtr<nsIRunnable> runnable =
+      NewRunnableMethod<StorensRefPtrPassByPtr<Image>, bool>(
+        this, &VideoFrameConverter::ProcessVideoFrame,
+        aChunk.mFrame.GetImage(), forceBlack);
+    mTaskQueue->Dispatch(runnable.forget());
+  }
+
+  void AddListener(VideoConverterListener* aListener)
+  {
+    MutexAutoLock lock(mMutex);
+
+    MOZ_ASSERT(!mListeners.Contains(aListener));
+    mListeners.AppendElement(aListener);
+  }
+
+  bool RemoveListener(VideoConverterListener* aListener)
+  {
+    MutexAutoLock lock(mMutex);
+
+    return mListeners.RemoveElement(aListener);
+  }
+
+  void Shutdown()
+  {
+    mTaskQueue->BeginShutdown();
+    mTaskQueue->AwaitShutdownAndIdle();
+  }
+
+protected:
+  virtual ~VideoFrameConverter()
+  {
+    MOZ_COUNT_DTOR(VideoFrameConverter);
+  }
+
+  void VideoFrameConverted(unsigned char* aVideoFrame,
+                           unsigned int aVideoFrameLength,
+                           unsigned short aWidth,
+                           unsigned short aHeight,
+                           VideoType aVideoType,
+                           uint64_t aCaptureTime)
+  {
+    MutexAutoLock lock(mMutex);
+
+    for (RefPtr<VideoConverterListener>& listener : mListeners) {
+      listener->OnVideoFrameConverted(aVideoFrame, aVideoFrameLength,
+                                      aWidth, aHeight, aVideoType, aCaptureTime);
+    }
+  }
+
+  void VideoFrameConverted(webrtc::I420VideoFrame& aVideoFrame)
+  {
+    MutexAutoLock lock(mMutex);
+
+    for (RefPtr<VideoConverterListener>& listener : mListeners) {
+      listener->OnVideoFrameConverted(aVideoFrame);
+    }
+  }
+
+  void ProcessVideoFrame(Image* aImage, bool aForceBlack)
+  {
+    --mLength; // Atomic
+    MOZ_ASSERT(mLength >= 0);
+
+    if (aForceBlack) {
+      IntSize size = aImage->GetSize();
+      uint32_t yPlaneLen = YSIZE(size.width, size.height);
+      uint32_t cbcrPlaneLen = 2 * CRSIZE(size.width, size.height);
+      uint32_t length = yPlaneLen + cbcrPlaneLen;
+
+      // Send a black image.
+      auto pixelData = MakeUniqueFallible<uint8_t[]>(length);
+      if (pixelData) {
+        // YCrCb black = 0x10 0x80 0x80
+        memset(pixelData.get(), 0x10, yPlaneLen);
+        // Fill Cb/Cr planes
+        memset(pixelData.get() + yPlaneLen, 0x80, cbcrPlaneLen);
+
+        MOZ_MTLOG(ML_DEBUG, "Sending a black video frame");
+        VideoFrameConverted(pixelData.get(), length, size.width, size.height,
+                            mozilla::kVideoI420, 0);
+      }
+      return;
+    }
+
+    ImageFormat format = aImage->GetFormat();
+#ifdef WEBRTC_GONK
+    GrallocImage* nativeImage = aImage->AsGrallocImage();
+    if (nativeImage) {
+      android::sp<android::GraphicBuffer> graphicBuffer = nativeImage->GetGraphicBuffer();
+      int pixelFormat = graphicBuffer->getPixelFormat(); /* PixelFormat is an enum == int */
+      mozilla::VideoType destFormat;
+      switch (pixelFormat) {
+        case HAL_PIXEL_FORMAT_YV12:
+          // all android must support this
+          destFormat = mozilla::kVideoYV12;
+          break;
+        case GrallocImage::HAL_PIXEL_FORMAT_YCbCr_420_SP:
+          destFormat = mozilla::kVideoNV21;
+          break;
+        case GrallocImage::HAL_PIXEL_FORMAT_YCbCr_420_P:
+          destFormat = mozilla::kVideoI420;
+          break;
+        default:
+          // XXX Bug NNNNNNN
+          // use http://dxr.mozilla.org/mozilla-central/source/content/media/omx/I420ColorConverterHelper.cpp
+          // to convert unknown types (OEM-specific) to I420
+          MOZ_MTLOG(ML_ERROR, "Un-handled GRALLOC buffer type:" << pixelFormat);
+          MOZ_CRASH();
+      }
+      void *basePtr;
+      graphicBuffer->lock(android::GraphicBuffer::USAGE_SW_READ_MASK, &basePtr);
+      uint32_t width = graphicBuffer->getWidth();
+      uint32_t height = graphicBuffer->getHeight();
+      // XXX gralloc buffer's width and stride could be different depends on implementations.
+
+      if (destFormat != mozilla::kVideoI420) {
+        unsigned char *video_frame = static_cast<unsigned char*>(basePtr);
+        webrtc::I420VideoFrame i420_frame;
+        int stride_y = width;
+        int stride_uv = (width + 1) / 2;
+        int target_width = width;
+        int target_height = height;
+        if (i420_frame.CreateEmptyFrame(target_width,
+                                        abs(target_height),
+                                        stride_y,
+                                        stride_uv, stride_uv) < 0) {
+          MOZ_ASSERT(false, "Can't allocate empty i420frame");
+          return;
+        }
+        webrtc::VideoType commonVideoType =
+          webrtc::RawVideoTypeToCommonVideoVideoType(
+            static_cast<webrtc::RawVideoType>((int)destFormat));
+        if (ConvertToI420(commonVideoType, video_frame, 0, 0, width, height,
+                          I420SIZE(width, height), webrtc::kVideoRotation_0,
+                          &i420_frame)) {
+          MOZ_ASSERT(false, "Can't convert video type for sending to I420");
+          return;
+        }
+        i420_frame.set_ntp_time_ms(0);
+        VideoFrameConverted(i420_frame);
+      } else {
+        VideoFrameConverted(static_cast<unsigned char*>(basePtr),
+                            I420SIZE(width, height),
+                            width,
+                            height,
+                            destFormat, 0);
+      }
+      graphicBuffer->unlock();
+      return;
+    } else
+#endif
+    if (format == ImageFormat::PLANAR_YCBCR) {
+      // Cast away constness b/c some of the accessors are non-const
+      PlanarYCbCrImage* yuv = const_cast<PlanarYCbCrImage *>(
+          static_cast<const PlanarYCbCrImage *>(aImage));
+
+      const PlanarYCbCrData *data = yuv->GetData();
+      if (data) {
+        uint8_t *y = data->mYChannel;
+        uint8_t *cb = data->mCbChannel;
+        uint8_t *cr = data->mCrChannel;
+        int32_t yStride = data->mYStride;
+        int32_t cbCrStride = data->mCbCrStride;
+        uint32_t width = yuv->GetSize().width;
+        uint32_t height = yuv->GetSize().height;
+
+        webrtc::I420VideoFrame i420_frame;
+        int rv = i420_frame.CreateFrame(y, cb, cr, width, height,
+                                        yStride, cbCrStride, cbCrStride,
+                                        webrtc::kVideoRotation_0);
+        if (rv != 0) {
+          NS_ERROR("Creating an I420 frame failed");
+          return;
+        }
+
+        MOZ_MTLOG(ML_DEBUG, "Sending an I420 video frame");
+        VideoFrameConverted(i420_frame);
+        return;
+      }
+    }
+
+    RefPtr<SourceSurface> surf = aImage->GetAsSourceSurface();
+    if (!surf) {
+      MOZ_MTLOG(ML_ERROR, "Getting surface from " << Stringify(format) << " image failed");
+      return;
+    }
+
+    RefPtr<DataSourceSurface> data = surf->GetDataSurface();
+    if (!data) {
+      MOZ_MTLOG(ML_ERROR, "Getting data surface from " << Stringify(format)
+                          << " image with " << Stringify(surf->GetType()) << "("
+                          << Stringify(surf->GetFormat()) << ") surface failed");
+      return;
+    }
+
+    IntSize size = aImage->GetSize();
+    int half_width = (size.width + 1) >> 1;
+    int half_height = (size.height + 1) >> 1;
+    int c_size = half_width * half_height;
+    int buffer_size = YSIZE(size.width, size.height) + 2 * c_size;
+    auto yuv_scoped = MakeUniqueFallible<uint8[]>(buffer_size);
+    if (!yuv_scoped) {
+      return;
+    }
+    uint8* yuv = yuv_scoped.get();
+
+    DataSourceSurface::ScopedMap map(data, DataSourceSurface::READ);
+    if (!map.IsMapped()) {
+      MOZ_MTLOG(ML_ERROR, "Reading DataSourceSurface from " << Stringify(format)
+                          << " image with " << Stringify(surf->GetType()) << "("
+                          << Stringify(surf->GetFormat()) << ") surface failed");
+      return;
+    }
+
+    int rv;
+    int cb_offset = YSIZE(size.width, size.height);
+    int cr_offset = cb_offset + c_size;
+    switch (surf->GetFormat()) {
+      case SurfaceFormat::B8G8R8A8:
+      case SurfaceFormat::B8G8R8X8:
+        rv = libyuv::ARGBToI420(static_cast<uint8*>(map.GetData()),
+                                map.GetStride(),
+                                yuv, size.width,
+                                yuv + cb_offset, half_width,
+                                yuv + cr_offset, half_width,
+                                size.width, size.height);
+        break;
+      case SurfaceFormat::R5G6B5_UINT16:
+        rv = libyuv::RGB565ToI420(static_cast<uint8*>(map.GetData()),
+                                  map.GetStride(),
+                                  yuv, size.width,
+                                  yuv + cb_offset, half_width,
+                                  yuv + cr_offset, half_width,
+                                  size.width, size.height);
+        break;
+      default:
+        MOZ_MTLOG(ML_ERROR, "Unsupported RGB video format" << Stringify(surf->GetFormat()));
+        MOZ_ASSERT(PR_FALSE);
+        return;
+    }
+    if (rv != 0) {
+      MOZ_MTLOG(ML_ERROR, Stringify(surf->GetFormat()) << " to I420 conversion failed");
+      return;
+    }
+    MOZ_MTLOG(ML_DEBUG, "Sending an I420 video frame converted from " <<
+                        Stringify(surf->GetFormat()));
+    VideoFrameConverted(yuv, buffer_size, size.width, size.height, mozilla::kVideoI420, 0);
+  }
+
+  Atomic<int32_t, Relaxed> mLength;
+  RefPtr<TaskQueue> mTaskQueue;
+
+  // Written and read from the queueing thread (normally MSG).
+  int32_t last_img_; // serial number of last Image
+  bool disabled_frame_sent_; // If a black frame has been sent after disabling.
+#ifdef DEBUG
+  uint32_t mThrottleCount;
+  uint32_t mThrottleRecord;
+#endif
+
+  // mMutex guards the below variables.
+  Mutex mMutex;
+  nsTArray<RefPtr<VideoConverterListener>> mListeners;
+};
+#endif
+
+// An async inserter for audio data, to avoid running audio codec encoders
+// on the MSG/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(AudioSessionConduit *aConduit)
+    : mConduit(aConduit)
+  {
+    MOZ_ASSERT(mConduit);
+    MOZ_COUNT_CTOR(AudioProxyThread);
+
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+    // Use only 1 thread; also forces FIFO operation
+    // We could use multiple threads, but that may be dicier with the webrtc.org
+    // code.  If so we'd need to use TaskQueues like the videoframe converter
+    RefPtr<SharedThreadPool> pool =
+      SharedThreadPool::Get(NS_LITERAL_CSTRING("AudioProxy"), 1);
+
+    mThread = pool.get();
+#else
+    nsCOMPtr<nsIThread> thread;
+    if (!NS_WARN_IF(NS_FAILED(NS_NewNamedThread("AudioProxy", getter_AddRefs(thread))))) {
+      mThread = thread;
+    }
+#endif
+  }
+
+  // called on mThread
+  void InternalProcessAudioChunk(
+    TrackRate rate,
+    AudioChunk& chunk,
+    bool enabled) {
+
+    // 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).
+    uint32_t outputChannels = chunk.ChannelCount() == 1 ? 1 : 2;
+    const int16_t* samples = nullptr;
+    UniquePtr<int16_t[]> convertedSamples;
+
+    // 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 (enabled && outputChannels == 1 && chunk.mBufferFormat == AUDIO_FORMAT_S16) {
+      samples = chunk.ChannelData<int16_t>().Elements()[0];
+    } else {
+      convertedSamples = MakeUnique<int16_t[]>(chunk.mDuration * outputChannels);
+
+      if (!enabled || chunk.mBufferFormat == AUDIO_FORMAT_SILENCE) {
+        PodZero(convertedSamples.get(), chunk.mDuration * outputChannels);
+      } else if (chunk.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
+        DownmixAndInterleave(chunk.ChannelData<float>(),
+                             chunk.mDuration, chunk.mVolume, outputChannels,
+                             convertedSamples.get());
+      } else if (chunk.mBufferFormat == AUDIO_FORMAT_S16) {
+        DownmixAndInterleave(chunk.ChannelData<int16_t>(),
+                             chunk.mDuration, chunk.mVolume, outputChannels,
+                             convertedSamples.get());
+      }
+      samples = convertedSamples.get();
+    }
+
+    MOZ_ASSERT(!(rate%100)); // rate should be a multiple of 100
+
+    // Check if the rate or the number of channels has changed since the last time
+    // we came through. I realize it may be overkill to check if the rate has
+    // changed, but I believe it is possible (e.g. if we change sources) and it
+    // costs us very little to handle this case.
+
+    uint32_t audio_10ms = rate / 100;
+
+    if (!packetizer_ ||
+        packetizer_->PacketSize() != audio_10ms ||
+        packetizer_->Channels() != outputChannels) {
+      // It's ok to drop the audio still in the packetizer here.
+      packetizer_ = new AudioPacketizer<int16_t, int16_t>(audio_10ms, outputChannels);
+    }
+
+    packetizer_->Input(samples, chunk.mDuration);
+
+    while (packetizer_->PacketsAvailable()) {
+      uint32_t samplesPerPacket = packetizer_->PacketSize() *
+                                  packetizer_->Channels();
+      // We know that webrtc.org's code going to copy the samples down the line,
+      // so we can just use a stack buffer here instead of malloc-ing.
+      int16_t packet[AUDIO_SAMPLE_BUFFER_MAX];
+
+      packetizer_->Output(packet);
+      mConduit->SendAudioFrame(packet, samplesPerPacket, rate, 0);
+    }
+  }
+
+  void QueueAudioChunk(TrackRate rate, AudioChunk& chunk, bool enabled)
+  {
+    RUN_ON_THREAD(mThread,
+                  WrapRunnable(RefPtr<AudioProxyThread>(this),
+                               &AudioProxyThread::InternalProcessAudioChunk,
+                               rate, chunk, enabled),
+                  NS_DISPATCH_NORMAL);
+  }
+
+protected:
+  virtual ~AudioProxyThread()
+  {
+    // Conduits must be released on MainThread, and we might have the last reference
+    // We don't need to worry about runnables still trying to access the conduit, since
+    // the runnables hold a ref to AudioProxyThread.
+    NS_ReleaseOnMainThread(mConduit.forget());
+    MOZ_COUNT_DTOR(AudioProxyThread);
+  }
+
+  RefPtr<AudioSessionConduit> mConduit;
+  nsCOMPtr<nsIEventTarget> mThread;
+  // Only accessed on mThread
+  nsAutoPtr<AudioPacketizer<int16_t, int16_t>> packetizer_;
+};
+
+static char kDTLSExporterLabel[] = "EXTRACTOR-dtls_srtp";
+
+MediaPipeline::MediaPipeline(const std::string& pc,
+                             Direction direction,
+                             nsCOMPtr<nsIEventTarget> main_thread,
+                             nsCOMPtr<nsIEventTarget> sts_thread,
+                             const std::string& track_id,
+                             int level,
+                             RefPtr<MediaSessionConduit> conduit,
+                             RefPtr<TransportFlow> rtp_transport,
+                             RefPtr<TransportFlow> rtcp_transport,
+                             nsAutoPtr<MediaPipelineFilter> filter)
+  : direction_(direction),
+    track_id_(track_id),
+    level_(level),
+    conduit_(conduit),
+    rtp_(rtp_transport, rtcp_transport ? RTP : MUX),
+    rtcp_(rtcp_transport ? rtcp_transport : rtp_transport,
+          rtcp_transport ? RTCP : MUX),
+    main_thread_(main_thread),
+    sts_thread_(sts_thread),
+    rtp_packets_sent_(0),
+    rtcp_packets_sent_(0),
+    rtp_packets_received_(0),
+    rtcp_packets_received_(0),
+    rtp_bytes_sent_(0),
+    rtp_bytes_received_(0),
+    pc_(pc),
+    description_(),
+    filter_(filter),
+    rtp_parser_(webrtc::RtpHeaderParser::Create()) {
+  // To indicate rtcp-mux rtcp_transport should be nullptr.
+  // Therefore it's an error to send in the same flow for
+  // both rtp and rtcp.
+  MOZ_ASSERT(rtp_transport != rtcp_transport);
+
+  // PipelineTransport() will access this->sts_thread_; moved here for safety
+  transport_ = new PipelineTransport(this);
+}
+
+MediaPipeline::~MediaPipeline() {
+  ASSERT_ON_THREAD(main_thread_);
+  MOZ_MTLOG(ML_INFO, "Destroying MediaPipeline: " << description_);
+}
+
+nsresult MediaPipeline::Init() {
+  ASSERT_ON_THREAD(main_thread_);
+
+  if (direction_ == RECEIVE) {
+    conduit_->SetReceiverTransport(transport_);
+  } else {
+    conduit_->SetTransmitterTransport(transport_);
+  }
+
+  RUN_ON_THREAD(sts_thread_,
+                WrapRunnable(
+                    RefPtr<MediaPipeline>(this),
+                    &MediaPipeline::Init_s),
+                NS_DISPATCH_NORMAL);
+
+  return NS_OK;
+}
+
+nsresult MediaPipeline::Init_s() {
+  ASSERT_ON_THREAD(sts_thread_);
+
+  return AttachTransport_s();
+}
+
+
+// Disconnect us from the transport so that we can cleanly destruct the
+// pipeline on the main thread.  ShutdownMedia_m() must have already been
+// called
+void
+MediaPipeline::DetachTransport_s()
+{
+  ASSERT_ON_THREAD(sts_thread_);
+
+  disconnect_all();
+  transport_->Detach();
+  rtp_.Detach();
+  rtcp_.Detach();
+}
+
+nsresult
+MediaPipeline::AttachTransport_s()
+{
+  ASSERT_ON_THREAD(sts_thread_);
+  nsresult res;
+  MOZ_ASSERT(rtp_.transport_);
+  MOZ_ASSERT(rtcp_.transport_);
+  res = ConnectTransport_s(rtp_);
+  if (NS_FAILED(res)) {
+    return res;
+  }
+
+  if (rtcp_.transport_ != rtp_.transport_) {
+    res = ConnectTransport_s(rtcp_);
+    if (NS_FAILED(res)) {
+      return res;
+    }
+  }
+
+  transport_->Attach(this);
+
+  return NS_OK;
+}
+
+void
+MediaPipeline::UpdateTransport_m(int level,
+                                 RefPtr<TransportFlow> rtp_transport,
+                                 RefPtr<TransportFlow> rtcp_transport,
+                                 nsAutoPtr<MediaPipelineFilter> filter)
+{
+  RUN_ON_THREAD(sts_thread_,
+                WrapRunnable(
+                    this,
+                    &MediaPipeline::UpdateTransport_s,
+                    level,
+                    rtp_transport,
+                    rtcp_transport,
+                    filter),
+                NS_DISPATCH_NORMAL);
+}
+
+void
+MediaPipeline::UpdateTransport_s(int level,
+                                 RefPtr<TransportFlow> rtp_transport,
+                                 RefPtr<TransportFlow> rtcp_transport,
+                                 nsAutoPtr<MediaPipelineFilter> filter)
+{
+  bool rtcp_mux = false;
+  if (!rtcp_transport) {
+    rtcp_transport = rtp_transport;
+    rtcp_mux = true;
+  }
+
+  if ((rtp_transport != rtp_.transport_) ||
+      (rtcp_transport != rtcp_.transport_)) {
+    DetachTransport_s();
+    rtp_ = TransportInfo(rtp_transport, rtcp_mux ? MUX : RTP);
+    rtcp_ = TransportInfo(rtcp_transport, rtcp_mux ? MUX : RTCP);
+    AttachTransport_s();
+  }
+
+  level_ = level;
+
+  if (filter_ && filter) {
+    // Use the new filter, but don't forget any remote SSRCs that we've learned
+    // by receiving traffic.
+    filter_->Update(*filter);
+  } else {
+    filter_ = filter;
+  }
+}
+
+void
+MediaPipeline::SelectSsrc_m(size_t ssrc_index)
+{
+  RUN_ON_THREAD(sts_thread_,
+                WrapRunnable(
+                    this,
+                    &MediaPipeline::SelectSsrc_s,
+                    ssrc_index),
+                NS_DISPATCH_NORMAL);
+}
+
+void
+MediaPipeline::SelectSsrc_s(size_t ssrc_index)
+{
+  filter_ = new MediaPipelineFilter;
+  if (ssrc_index < ssrcs_received_.size()) {
+    filter_->AddRemoteSSRC(ssrcs_received_[ssrc_index]);
+  } else {
+    MOZ_MTLOG(ML_WARNING, "SelectSsrc called with " << ssrc_index << " but we "
+                          << "have only seen " << ssrcs_received_.size()
+                          << " ssrcs");
+  }
+}
+
+void MediaPipeline::StateChange(TransportFlow *flow, TransportLayer::State state) {
+  TransportInfo* info = GetTransportInfo_s(flow);
+  MOZ_ASSERT(info);
+
+  if (state == TransportLayer::TS_OPEN) {
+    MOZ_MTLOG(ML_INFO, "Flow is ready");
+    TransportReady_s(*info);
+  } else if (state == TransportLayer::TS_CLOSED ||
+             state == TransportLayer::TS_ERROR) {
+    TransportFailed_s(*info);
+  }
+}
+
+static bool MakeRtpTypeToStringArray(const char** array) {
+  static const char* RTP_str = "RTP";
+  static const char* RTCP_str = "RTCP";
+  static const char* MUX_str = "RTP/RTCP mux";
+  array[MediaPipeline::RTP] = RTP_str;
+  array[MediaPipeline::RTCP] = RTCP_str;
+  array[MediaPipeline::MUX] = MUX_str;
+  return true;
+}
+
+static const char* ToString(MediaPipeline::RtpType type) {
+  static const char* array[(int)MediaPipeline::MAX_RTP_TYPE] = {nullptr};
+  // Dummy variable to cause init to happen only on first call
+  static bool dummy = MakeRtpTypeToStringArray(array);
+  (void)dummy;
+  return array[type];
+}
+
+nsresult MediaPipeline::TransportReady_s(TransportInfo &info) {
+  MOZ_ASSERT(!description_.empty());
+
+  // TODO(ekr@rtfm.com): implement some kind of notification on
+  // failure. bug 852665.
+  if (info.state_ != MP_CONNECTING) {
+    MOZ_MTLOG(ML_ERROR, "Transport ready for flow in wrong state:" <<
+              description_ << ": " << ToString(info.type_));
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_MTLOG(ML_INFO, "Transport ready for pipeline " <<
+            static_cast<void *>(this) << " flow " << description_ << ": " <<
+            ToString(info.type_));
+
+  // TODO(bcampen@mozilla.com): Should we disconnect from the flow on failure?
+  nsresult res;
+
+  // Now instantiate the SRTP objects
+  TransportLayerDtls *dtls = static_cast<TransportLayerDtls *>(
+      info.transport_->GetLayer(TransportLayerDtls::ID()));
+  MOZ_ASSERT(dtls);  // DTLS is mandatory
+
+  uint16_t cipher_suite;
+  res = dtls->GetSrtpCipher(&cipher_suite);
+  if (NS_FAILED(res)) {
+    MOZ_MTLOG(ML_ERROR, "Failed to negotiate DTLS-SRTP. This is an error");
+    info.state_ = MP_CLOSED;
+    UpdateRtcpMuxState(info);
+    return res;
+  }
+
+  // SRTP Key Exporter as per RFC 5764 S 4.2
+  unsigned char srtp_block[SRTP_TOTAL_KEY_LENGTH * 2];
+  res = dtls->ExportKeyingMaterial(kDTLSExporterLabel, false, "",
+                                   srtp_block, sizeof(srtp_block));
+  if (NS_FAILED(res)) {
+    MOZ_MTLOG(ML_ERROR, "Failed to compute DTLS-SRTP keys. This is an error");
+    info.state_ = MP_CLOSED;
+    UpdateRtcpMuxState(info);
+    MOZ_CRASH();  // TODO: Remove once we have enough field experience to
+                  // know it doesn't happen. bug 798797. Note that the
+                  // code after this never executes.
+    return res;
+  }
+
+  // Slice and dice as per RFC 5764 S 4.2
+  unsigned char client_write_key[SRTP_TOTAL_KEY_LENGTH];
+  unsigned char server_write_key[SRTP_TOTAL_KEY_LENGTH];
+  int offset = 0;
+  memcpy(client_write_key, srtp_block + offset, SRTP_MASTER_KEY_LENGTH);
+  offset += SRTP_MASTER_KEY_LENGTH;
+  memcpy(server_write_key, srtp_block + offset, SRTP_MASTER_KEY_LENGTH);
+  offset += SRTP_MASTER_KEY_LENGTH;
+  memcpy(client_write_key + SRTP_MASTER_KEY_LENGTH,
+         srtp_block + offset, SRTP_MASTER_SALT_LENGTH);
+  offset += SRTP_MASTER_SALT_LENGTH;
+  memcpy(server_write_key + SRTP_MASTER_KEY_LENGTH,
+         srtp_block + offset, SRTP_MASTER_SALT_LENGTH);
+  offset += SRTP_MASTER_SALT_LENGTH;
+  MOZ_ASSERT(offset == sizeof(srtp_block));
+
+  unsigned char *write_key;
+  unsigned char *read_key;
+
+  if (dtls->role() == TransportLayerDtls::CLIENT) {
+    write_key = client_write_key;
+    read_key = server_write_key;
+  } else {
+    write_key = server_write_key;
+    read_key = client_write_key;
+  }
+
+  MOZ_ASSERT(!info.send_srtp_ && !info.recv_srtp_);
+  info.send_srtp_ = SrtpFlow::Create(cipher_suite, false, write_key,
+                                     SRTP_TOTAL_KEY_LENGTH);
+  info.recv_srtp_ = SrtpFlow::Create(cipher_suite, true, read_key,
+                                     SRTP_TOTAL_KEY_LENGTH);
+  if (!info.send_srtp_ || !info.recv_srtp_) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create SRTP flow for "
+              << ToString(info.type_));
+    info.state_ = MP_CLOSED;
+    UpdateRtcpMuxState(info);
+    return NS_ERROR_FAILURE;
+  }
+
+    MOZ_MTLOG(ML_INFO, "Listening for " << ToString(info.type_)
+                       << " packets received on " <<
+                       static_cast<void *>(dtls->downward()));
+
+  switch (info.type_) {
+    case RTP:
+      dtls->downward()->SignalPacketReceived.connect(
+          this,
+          &MediaPipeline::RtpPacketReceived);
+      break;
+    case RTCP:
+      dtls->downward()->SignalPacketReceived.connect(
+          this,
+          &MediaPipeline::RtcpPacketReceived);
+      break;
+    case MUX:
+      dtls->downward()->SignalPacketReceived.connect(
+          this,
+          &MediaPipeline::PacketReceived);
+      break;
+    default:
+      MOZ_CRASH();
+  }
+
+  info.state_ = MP_OPEN;
+  UpdateRtcpMuxState(info);
+  return NS_OK;
+}
+
+nsresult MediaPipeline::TransportFailed_s(TransportInfo &info) {
+  ASSERT_ON_THREAD(sts_thread_);
+
+  info.state_ = MP_CLOSED;
+  UpdateRtcpMuxState(info);
+
+  MOZ_MTLOG(ML_INFO, "Transport closed for flow " << ToString(info.type_));
+
+  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.
+
+  return NS_OK;
+}
+
+void MediaPipeline::UpdateRtcpMuxState(TransportInfo &info) {
+  if (info.type_ == MUX) {
+    if (info.transport_ == rtcp_.transport_) {
+      rtcp_.state_ = info.state_;
+      if (!rtcp_.send_srtp_) {
+        rtcp_.send_srtp_ = info.send_srtp_;
+        rtcp_.recv_srtp_ = info.recv_srtp_;
+      }
+    }
+  }
+}
+
+nsresult MediaPipeline::SendPacket(TransportFlow *flow, const void *data,
+                                   int len) {
+  ASSERT_ON_THREAD(sts_thread_);
+
+  // Note that we bypass the DTLS layer here
+  TransportLayerDtls *dtls = static_cast<TransportLayerDtls *>(
+      flow->GetLayer(TransportLayerDtls::ID()));
+  MOZ_ASSERT(dtls);
+
+  TransportResult res = dtls->downward()->
+      SendPacket(static_cast<const unsigned char *>(data), len);
+
+  if (res != len) {
+    // Ignore blocking indications
+    if (res == TE_WOULDBLOCK)
+      return NS_OK;
+
+    MOZ_MTLOG(ML_ERROR, "Failed write on stream " << description_);
+    return NS_BASE_STREAM_CLOSED;
+  }
+
+  return NS_OK;
+}
+
+void MediaPipeline::increment_rtp_packets_sent(int32_t bytes) {
+  ++rtp_packets_sent_;
+  rtp_bytes_sent_ += bytes;
+
+  if (!(rtp_packets_sent_ % 100)) {
+    MOZ_MTLOG(ML_INFO, "RTP sent packet count for " << description_
+              << " Pipeline " << static_cast<void *>(this)
+              << " Flow : " << static_cast<void *>(rtp_.transport_)
+              << ": " << rtp_packets_sent_
+              << " (" << rtp_bytes_sent_ << " bytes)");
+  }
+}
+
+void MediaPipeline::increment_rtcp_packets_sent() {
+  ++rtcp_packets_sent_;
+  if (!(rtcp_packets_sent_ % 100)) {
+    MOZ_MTLOG(ML_INFO, "RTCP sent packet count for " << description_
+              << " Pipeline " << static_cast<void *>(this)
+              << " Flow : " << static_cast<void *>(rtcp_.transport_)
+              << ": " << rtcp_packets_sent_);
+  }
+}
+
+void MediaPipeline::increment_rtp_packets_received(int32_t bytes) {
+  ++rtp_packets_received_;
+  rtp_bytes_received_ += bytes;
+  if (!(rtp_packets_received_ % 100)) {
+    MOZ_MTLOG(ML_INFO, "RTP received packet count for " << description_
+              << " Pipeline " << static_cast<void *>(this)
+              << " Flow : " << static_cast<void *>(rtp_.transport_)
+              << ": " << rtp_packets_received_
+              << " (" << rtp_bytes_received_ << " bytes)");
+  }
+}
+
+void MediaPipeline::increment_rtcp_packets_received() {
+  ++rtcp_packets_received_;
+  if (!(rtcp_packets_received_ % 100)) {
+    MOZ_MTLOG(ML_INFO, "RTCP received packet count for " << description_
+              << " Pipeline " << static_cast<void *>(this)
+              << " Flow : " << static_cast<void *>(rtcp_.transport_)
+              << ": " << rtcp_packets_received_);
+  }
+}
+
+void MediaPipeline::RtpPacketReceived(TransportLayer *layer,
+                                      const unsigned char *data,
+                                      size_t len) {
+  if (!transport_->pipeline()) {
+    MOZ_MTLOG(ML_ERROR, "Discarding incoming packet; transport disconnected");
+    return;
+  }
+
+  if (!conduit_) {
+    MOZ_MTLOG(ML_DEBUG, "Discarding incoming packet; media disconnected");
+    return;
+  }
+
+  if (rtp_.state_ != MP_OPEN) {
+    MOZ_MTLOG(ML_ERROR, "Discarding incoming packet; pipeline not open");
+    return;
+  }
+
+  if (rtp_.transport_->state() != TransportLayer::TS_OPEN) {
+    MOZ_MTLOG(ML_ERROR, "Discarding incoming packet; transport not open");
+    return;
+  }
+
+  // This should never happen.
+  MOZ_ASSERT(rtp_.recv_srtp_);
+
+  if (direction_ == TRANSMIT) {
+    return;
+  }
+
+  if (!len) {
+    return;
+  }
+
+  // Filter out everything but RTP/RTCP
+  if (data[0] < 128 || data[0] > 191) {
+    return;
+  }
+
+  webrtc::RTPHeader header;
+  if (!rtp_parser_->Parse(data, len, &header)) {
+    return;
+  }
+
+  if (std::find(ssrcs_received_.begin(), ssrcs_received_.end(), header.ssrc) ==
+      ssrcs_received_.end()) {
+    ssrcs_received_.push_back(header.ssrc);
+  }
+
+  if (filter_ && !filter_->Filter(header)) {
+    return;
+  }
+
+  // Make a copy rather than cast away constness
+  auto inner_data = MakeUnique<unsigned char[]>(len);
+  memcpy(inner_data.get(), data, len);
+  int out_len = 0;
+  nsresult res = rtp_.recv_srtp_->UnprotectRtp(inner_data.get(),
+                                               len, len, &out_len);
+  if (!NS_SUCCEEDED(res)) {
+    char tmp[16];
+
+    SprintfLiteral(tmp, "%.2x %.2x %.2x %.2x",
+                   inner_data[0],
+                   inner_data[1],
+                   inner_data[2],
+                   inner_data[3]);
+
+    MOZ_MTLOG(ML_NOTICE, "Error unprotecting RTP in " << description_
+              << "len= " << len << "[" << tmp << "...]");
+
+    return;
+  }
+  MOZ_MTLOG(ML_DEBUG, description_ << " received RTP packet.");
+  increment_rtp_packets_received(out_len);
+
+  (void)conduit_->ReceivedRTPPacket(inner_data.get(), out_len);  // Ignore error codes
+}
+
+void MediaPipeline::RtcpPacketReceived(TransportLayer *layer,
+                                       const unsigned char *data,
+                                       size_t len) {
+  if (!transport_->pipeline()) {
+    MOZ_MTLOG(ML_DEBUG, "Discarding incoming packet; transport disconnected");
+    return;
+  }
+
+  if (!conduit_) {
+    MOZ_MTLOG(ML_DEBUG, "Discarding incoming packet; media disconnected");
+    return;
+  }
+
+  if (rtcp_.state_ != MP_OPEN) {
+    MOZ_MTLOG(ML_DEBUG, "Discarding incoming packet; pipeline not open");
+    return;
+  }
+
+  if (rtcp_.transport_->state() != TransportLayer::TS_OPEN) {
+    MOZ_MTLOG(ML_ERROR, "Discarding incoming packet; transport not open");
+    return;
+  }
+
+  if (!len) {
+    return;
+  }
+
+  // Filter out everything but RTP/RTCP
+  if (data[0] < 128 || data[0] > 191) {
+    return;
+  }
+
+  // We do not filter RTCP for send pipelines, since the webrtc.org code for
+  // senders already has logic to ignore RRs that do not apply.
+  // TODO bug 1279153: remove SR check for reduced size RTCP
+  if (filter_ && direction_ == RECEIVE) {
+    if (!filter_->FilterSenderReport(data, len)) {
+      MOZ_MTLOG(ML_NOTICE, "Dropping incoming RTCP packet; filtered out");
+      return;
+    }
+  }
+
+  // Make a copy rather than cast away constness
+  auto inner_data = MakeUnique<unsigned char[]>(len);
+  memcpy(inner_data.get(), data, len);
+  int out_len;
+
+  nsresult res = rtcp_.recv_srtp_->UnprotectRtcp(inner_data.get(),
+                                                 len,
+                                                 len,
+                                                 &out_len);
+
+  if (!NS_SUCCEEDED(res))
+    return;
+
+  MOZ_MTLOG(ML_DEBUG, description_ << " received RTCP packet.");
+  increment_rtcp_packets_received();
+
+  MOZ_ASSERT(rtcp_.recv_srtp_);  // This should never happen
+
+  (void)conduit_->ReceivedRTCPPacket(inner_data.get(), out_len);  // Ignore error codes
+}
+
+bool MediaPipeline::IsRtp(const unsigned char *data, size_t len) {
+  if (len < 2)
+    return false;
+
+  // Check if this is a RTCP packet. Logic based on the types listed in
+  // media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_utility.cc
+
+  // Anything outside this range is RTP.
+  if ((data[1] < 192) || (data[1] > 207))
+    return true;
+
+  if (data[1] == 192)  // FIR
+    return false;
+
+  if (data[1] == 193)  // NACK, but could also be RTP. This makes us sad
+    return true;       // but it's how webrtc.org behaves.
+
+  if (data[1] == 194)
+    return true;
+
+  if (data[1] == 195)  // IJ.
+    return false;
+
+  if ((data[1] > 195) && (data[1] < 200))  // the > 195 is redundant
+    return true;
+
+  if ((data[1] >= 200) && (data[1] <= 207))  // SR, RR, SDES, BYE,
+    return false;                            // APP, RTPFB, PSFB, XR
+
+  MOZ_ASSERT(false);  // Not reached, belt and suspenders.
+  return true;
+}
+
+void MediaPipeline::PacketReceived(TransportLayer *layer,
+                                   const unsigned char *data,
+                                   size_t len) {
+  if (!transport_->pipeline()) {
+    MOZ_MTLOG(ML_DEBUG, "Discarding incoming packet; transport disconnected");
+    return;
+  }
+
+  if (IsRtp(data, len)) {
+    RtpPacketReceived(layer, data, len);
+  } else {
+    RtcpPacketReceived(layer, data, len);
+  }
+}
+
+class MediaPipelineTransmit::PipelineListener
+  : public DirectMediaStreamTrackListener
+{
+friend class MediaPipelineTransmit;
+public:
+  explicit PipelineListener(const RefPtr<MediaSessionConduit>& conduit)
+    : conduit_(conduit),
+      track_id_(TRACK_INVALID),
+      mMutex("MediaPipelineTransmit::PipelineListener"),
+      track_id_external_(TRACK_INVALID),
+      active_(false),
+      enabled_(false),
+      direct_connect_(false)
+  {
+  }
+
+  ~PipelineListener()
+  {
+    if (!NS_IsMainThread()) {
+      // release conduit on mainthread.  Must use forget()!
+      nsresult rv = NS_DispatchToMainThread(new
+                                            ConduitDeleteEvent(conduit_.forget()));
+      MOZ_ASSERT(!NS_FAILED(rv),"Could not dispatch conduit shutdown to main");
+      if (NS_FAILED(rv)) {
+        MOZ_CRASH();
+      }
+    } else {
+      conduit_ = nullptr;
+    }
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+    if (converter_) {
+      converter_->Shutdown();
+    }
+#endif
+  }
+
+  // Dispatches setting the internal TrackID to TRACK_INVALID to the media
+  // graph thread to keep it in sync with other MediaStreamGraph operations
+  // like RemoveListener() and AddListener(). The TrackID will be updated on
+  // the next NewData() callback.
+  void UnsetTrackId(MediaStreamGraphImpl* graph);
+
+  void SetActive(bool active) { active_ = active; }
+  void SetEnabled(bool enabled) { enabled_ = enabled; }
+
+  // These are needed since nested classes don't have access to any particular
+  // instance of the parent
+  void SetAudioProxy(const RefPtr<AudioProxyThread>& proxy)
+  {
+    audio_processing_ = proxy;
+  }
+
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  void SetVideoFrameConverter(const RefPtr<VideoFrameConverter>& converter)
+  {
+    converter_ = converter;
+  }
+
+  void OnVideoFrameConverted(unsigned char* aVideoFrame,
+                             unsigned int aVideoFrameLength,
+                             unsigned short aWidth,
+                             unsigned short aHeight,
+                             VideoType aVideoType,
+                             uint64_t aCaptureTime)
+  {
+    MOZ_ASSERT(conduit_->type() == MediaSessionConduit::VIDEO);
+    static_cast<VideoSessionConduit*>(conduit_.get())->SendVideoFrame(
+      aVideoFrame, aVideoFrameLength, aWidth, aHeight, aVideoType, aCaptureTime);
+  }
+
+  void OnVideoFrameConverted(webrtc::I420VideoFrame& aVideoFrame)
+  {
+    MOZ_ASSERT(conduit_->type() == MediaSessionConduit::VIDEO);
+    static_cast<VideoSessionConduit*>(conduit_.get())->SendVideoFrame(aVideoFrame);
+  }
+#endif
+
+  // Implement MediaStreamTrackListener
+  void NotifyQueuedChanges(MediaStreamGraph* aGraph,
+                           StreamTime aTrackOffset,
+                           const MediaSegment& aQueuedMedia) override;
+
+  // Implement DirectMediaStreamTrackListener
+  void NotifyRealtimeTrackData(MediaStreamGraph* aGraph,
+                               StreamTime aTrackOffset,
+                               const MediaSegment& aMedia) override;
+  void NotifyDirectListenerInstalled(InstallationResult aResult) override;
+  void NotifyDirectListenerUninstalled() override;
+
+private:
+  void UnsetTrackIdImpl() {
+    MutexAutoLock lock(mMutex);
+    track_id_ = track_id_external_ = TRACK_INVALID;
+  }
+
+  void NewData(MediaStreamGraph* graph,
+               StreamTime offset,
+               const MediaSegment& media);
+
+  RefPtr<MediaSessionConduit> conduit_;
+  RefPtr<AudioProxyThread> audio_processing_;
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  RefPtr<VideoFrameConverter> converter_;
+#endif
+
+  // May be TRACK_INVALID until we see data from the track
+  TrackID track_id_; // this is the current TrackID this listener is attached to
+  Mutex mMutex;
+  // protected by mMutex
+  // May be TRACK_INVALID until we see data from the track
+  TrackID track_id_external_; // this is queried from other threads
+
+  // active is true if there is a transport to send on
+  mozilla::Atomic<bool> active_;
+  // enabled is true if the media access control permits sending
+  // actual content; when false you get black/silence
+  mozilla::Atomic<bool> enabled_;
+
+  // Written and read on the MediaStreamGraph thread
+  bool direct_connect_;
+};
+
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+// Implements VideoConverterListener for MediaPipeline.
+//
+// We pass converted frames on to MediaPipelineTransmit::PipelineListener
+// where they are further forwarded to VideoConduit.
+// MediaPipelineTransmit calls Detach() during shutdown to ensure there is
+// no cyclic dependencies between us and PipelineListener.
+class MediaPipelineTransmit::VideoFrameFeeder
+  : public VideoConverterListener
+{
+public:
+  explicit VideoFrameFeeder(const RefPtr<PipelineListener>& listener)
+    : listener_(listener),
+      mutex_("VideoFrameFeeder")
+  {
+    MOZ_COUNT_CTOR(VideoFrameFeeder);
+  }
+
+  void Detach()
+  {
+    MutexAutoLock lock(mutex_);
+
+    listener_ = nullptr;
+  }
+
+  void OnVideoFrameConverted(unsigned char* aVideoFrame,
+                             unsigned int aVideoFrameLength,
+                             unsigned short aWidth,
+                             unsigned short aHeight,
+                             VideoType aVideoType,
+                             uint64_t aCaptureTime) override
+  {
+    MutexAutoLock lock(mutex_);
+
+    if (!listener_) {
+      return;
+    }
+
+    listener_->OnVideoFrameConverted(aVideoFrame, aVideoFrameLength,
+                                     aWidth, aHeight, aVideoType, aCaptureTime);
+  }
+
+  void OnVideoFrameConverted(webrtc::I420VideoFrame& aVideoFrame) override
+  {
+    MutexAutoLock lock(mutex_);
+
+    if (!listener_) {
+      return;
+    }
+
+    listener_->OnVideoFrameConverted(aVideoFrame);
+  }
+
+protected:
+  virtual ~VideoFrameFeeder()
+  {
+    MOZ_COUNT_DTOR(VideoFrameFeeder);
+  }
+
+  RefPtr<PipelineListener> listener_;
+  Mutex mutex_;
+};
+#endif
+
+class MediaPipelineTransmit::PipelineVideoSink :
+  public MediaStreamVideoSink
+{
+public:
+  explicit PipelineVideoSink(const RefPtr<MediaSessionConduit>& conduit,
+                             MediaPipelineTransmit::PipelineListener* listener)
+    : conduit_(conduit)
+    , pipelineListener_(listener)
+  {
+  }
+
+  virtual void SetCurrentFrames(const VideoSegment& aSegment) override;
+  virtual void ClearFrames() override {}
+
+private:
+  ~PipelineVideoSink() {
+    // release conduit on mainthread.  Must use forget()!
+    nsresult rv = NS_DispatchToMainThread(new
+      ConduitDeleteEvent(conduit_.forget()));
+    MOZ_ASSERT(!NS_FAILED(rv),"Could not dispatch conduit shutdown to main");
+    if (NS_FAILED(rv)) {
+      MOZ_CRASH();
+    }
+  }
+  RefPtr<MediaSessionConduit> conduit_;
+  MediaPipelineTransmit::PipelineListener* pipelineListener_;
+};
+
+MediaPipelineTransmit::MediaPipelineTransmit(
+    const std::string& pc,
+    nsCOMPtr<nsIEventTarget> main_thread,
+    nsCOMPtr<nsIEventTarget> sts_thread,
+    dom::MediaStreamTrack* domtrack,
+    const std::string& track_id,
+    int level,
+    RefPtr<MediaSessionConduit> conduit,
+    RefPtr<TransportFlow> rtp_transport,
+    RefPtr<TransportFlow> rtcp_transport,
+    nsAutoPtr<MediaPipelineFilter> filter) :
+  MediaPipeline(pc, TRANSMIT, main_thread, sts_thread, track_id, level,
+                conduit, rtp_transport, rtcp_transport, filter),
+  listener_(new PipelineListener(conduit)),
+  video_sink_(new PipelineVideoSink(conduit, listener_)),
+  domtrack_(domtrack)
+{
+  if (!IsVideo()) {
+    audio_processing_ = MakeAndAddRef<AudioProxyThread>(static_cast<AudioSessionConduit*>(conduit.get()));
+    listener_->SetAudioProxy(audio_processing_);
+  }
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  else { // Video
+    // For video we send frames to an async VideoFrameConverter that calls
+    // back to a VideoFrameFeeder that feeds I420 frames to VideoConduit.
+
+    feeder_ = MakeAndAddRef<VideoFrameFeeder>(listener_);
+
+    converter_ = MakeAndAddRef<VideoFrameConverter>();
+    converter_->AddListener(feeder_);
+
+    listener_->SetVideoFrameConverter(converter_);
+  }
+#endif
+}
+
+MediaPipelineTransmit::~MediaPipelineTransmit()
+{
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  if (feeder_) {
+    feeder_->Detach();
+  }
+#endif
+}
+
+nsresult MediaPipelineTransmit::Init() {
+  AttachToTrack(track_id_);
+
+  return MediaPipeline::Init();
+}
+
+void MediaPipelineTransmit::AttachToTrack(const std::string& track_id) {
+  ASSERT_ON_THREAD(main_thread_);
+
+  description_ = pc_ + "| ";
+  description_ += conduit_->type() == MediaSessionConduit::AUDIO ?
+      "Transmit audio[" : "Transmit video[";
+  description_ += track_id;
+  description_ += "]";
+
+  // TODO(ekr@rtfm.com): Check for errors
+  MOZ_MTLOG(ML_DEBUG, "Attaching pipeline to track "
+            << static_cast<void *>(domtrack_) << " conduit type=" <<
+            (conduit_->type() == MediaSessionConduit::AUDIO ?"audio":"video"));
+
+  // Register the Listener directly with the source if we can.
+  // We also register it as a non-direct listener so we fall back to that
+  // if installing the direct listener fails. As a direct listener we get access
+  // to direct unqueued (and not resampled) data.
+  domtrack_->AddDirectListener(listener_);
+  domtrack_->AddListener(listener_);
+
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  domtrack_->AddDirectListener(video_sink_);
+#endif
+
+#ifndef MOZILLA_INTERNAL_API
+  // this enables the unit tests that can't fiddle with principals and the like
+  listener_->SetEnabled(true);
+#endif
+}
+
+bool
+MediaPipelineTransmit::IsVideo() const
+{
+  return !!domtrack_->AsVideoStreamTrack();
+}
+
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+void MediaPipelineTransmit::UpdateSinkIdentity_m(MediaStreamTrack* track,
+                                                 nsIPrincipal* principal,
+                                                 const PeerIdentity* sinkIdentity) {
+  ASSERT_ON_THREAD(main_thread_);
+
+  if (track != nullptr && track != domtrack_) {
+    // If a track is specified, then it might not be for this pipeline,
+    // since we receive notifications for all tracks on the PC.
+    // nullptr means that the PeerIdentity has changed and shall be applied
+    // to all tracks of the PC.
+    return;
+  }
+
+  bool enableTrack = principal->Subsumes(domtrack_->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 = domtrack_->GetPeerIdentity();
+    if (sinkIdentity && trackIdentity) {
+      enableTrack = (*sinkIdentity == *trackIdentity);
+    }
+  }
+
+  listener_->SetEnabled(enableTrack);
+}
+#endif
+
+void
+MediaPipelineTransmit::DetachMedia()
+{
+  ASSERT_ON_THREAD(main_thread_);
+  if (domtrack_) {
+    domtrack_->RemoveDirectListener(listener_);
+    domtrack_->RemoveListener(listener_);
+    domtrack_->RemoveDirectListener(video_sink_);
+    domtrack_ = nullptr;
+  }
+  // Let the listener be destroyed with the pipeline (or later).
+}
+
+nsresult MediaPipelineTransmit::TransportReady_s(TransportInfo &info) {
+  ASSERT_ON_THREAD(sts_thread_);
+  // Call base ready function.
+  MediaPipeline::TransportReady_s(info);
+
+  // Should not be set for a transmitter
+  if (&info == &rtp_) {
+    listener_->SetActive(true);
+  }
+
+  return NS_OK;
+}
+
+nsresult MediaPipelineTransmit::ReplaceTrack(MediaStreamTrack& domtrack) {
+  // MainThread, checked in calls we make
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  nsString nsTrackId;
+  domtrack.GetId(nsTrackId);
+  std::string track_id(NS_ConvertUTF16toUTF8(nsTrackId).get());
+#else
+  std::string track_id = domtrack.GetId();
+#endif
+  MOZ_MTLOG(ML_DEBUG, "Reattaching pipeline " << description_ << " to track "
+            << static_cast<void *>(&domtrack)
+            << " track " << track_id << " conduit type=" <<
+            (conduit_->type() == MediaSessionConduit::AUDIO ?"audio":"video"));
+
+  DetachMedia();
+  domtrack_ = &domtrack; // Detach clears it
+  // Unsets the track id after RemoveListener() takes effect.
+  listener_->UnsetTrackId(domtrack_->GraphImpl());
+  track_id_ = track_id;
+  AttachToTrack(track_id);
+  return NS_OK;
+}
+
+void MediaPipeline::DisconnectTransport_s(TransportInfo &info) {
+  MOZ_ASSERT(info.transport_);
+  ASSERT_ON_THREAD(sts_thread_);
+
+  info.transport_->SignalStateChange.disconnect(this);
+  // We do this even if we're a transmitter, since we are still possibly
+  // registered to receive RTCP.
+  TransportLayerDtls *dtls = static_cast<TransportLayerDtls *>(
+      info.transport_->GetLayer(TransportLayerDtls::ID()));
+  MOZ_ASSERT(dtls);  // DTLS is mandatory
+  MOZ_ASSERT(dtls->downward());
+  dtls->downward()->SignalPacketReceived.disconnect(this);
+}
+
+nsresult MediaPipeline::ConnectTransport_s(TransportInfo &info) {
+  MOZ_ASSERT(info.transport_);
+  ASSERT_ON_THREAD(sts_thread_);
+
+  // Look to see if the transport is ready
+  if (info.transport_->state() == TransportLayer::TS_OPEN) {
+    nsresult res = TransportReady_s(info);
+    if (NS_FAILED(res)) {
+      MOZ_MTLOG(ML_ERROR, "Error calling TransportReady(); res="
+                << static_cast<uint32_t>(res) << " in " << __FUNCTION__);
+      return res;
+    }
+  } else if (info.transport_->state() == TransportLayer::TS_ERROR) {
+    MOZ_MTLOG(ML_ERROR, ToString(info.type_)
+                        << "transport is already in error state");
+    TransportFailed_s(info);
+    return NS_ERROR_FAILURE;
+  }
+
+  info.transport_->SignalStateChange.connect(this,
+                                             &MediaPipeline::StateChange);
+
+  return NS_OK;
+}
+
+MediaPipeline::TransportInfo* MediaPipeline::GetTransportInfo_s(
+    TransportFlow *flow) {
+  ASSERT_ON_THREAD(sts_thread_);
+  if (flow == rtp_.transport_) {
+    return &rtp_;
+  }
+
+  if (flow == rtcp_.transport_) {
+    return &rtcp_;
+  }
+
+  return nullptr;
+}
+
+nsresult MediaPipeline::PipelineTransport::SendRtpPacket(
+    const void *data, int len) {
+
+    nsAutoPtr<DataBuffer> buf(new DataBuffer(static_cast<const uint8_t *>(data),
+                                             len, len + SRTP_MAX_EXPANSION));
+
+    RUN_ON_THREAD(sts_thread_,
+                  WrapRunnable(
+                      RefPtr<MediaPipeline::PipelineTransport>(this),
+                      &MediaPipeline::PipelineTransport::SendRtpRtcpPacket_s,
+                      buf, true),
+                  NS_DISPATCH_NORMAL);
+
+    return NS_OK;
+}
+
+nsresult MediaPipeline::PipelineTransport::SendRtpRtcpPacket_s(
+    nsAutoPtr<DataBuffer> data,
+    bool is_rtp) {
+
+  ASSERT_ON_THREAD(sts_thread_);
+  if (!pipeline_) {
+    return NS_OK;  // Detached
+  }
+  TransportInfo& transport = is_rtp ? pipeline_->rtp_ : pipeline_->rtcp_;
+
+  if (!transport.send_srtp_) {
+    MOZ_MTLOG(ML_DEBUG, "Couldn't write RTP/RTCP packet; SRTP not set up yet");
+    return NS_OK;
+  }
+
+  MOZ_ASSERT(transport.transport_);
+  NS_ENSURE_TRUE(transport.transport_, NS_ERROR_NULL_POINTER);
+
+  // libsrtp enciphers in place, so we need a big enough buffer.
+  MOZ_ASSERT(data->capacity() >= data->len() + SRTP_MAX_EXPANSION);
+
+  int out_len;
+  nsresult res;
+  if (is_rtp) {
+    res = transport.send_srtp_->ProtectRtp(data->data(),
+                                           data->len(),
+                                           data->capacity(),
+                                           &out_len);
+  } else {
+    res = transport.send_srtp_->ProtectRtcp(data->data(),
+                                            data->len(),
+                                            data->capacity(),
+                                            &out_len);
+  }
+  if (!NS_SUCCEEDED(res)) {
+    return res;
+  }
+
+  // paranoia; don't have uninitialized bytes included in data->len()
+  data->SetLength(out_len);
+
+  MOZ_MTLOG(ML_DEBUG, pipeline_->description_ << " sending " <<
+            (is_rtp ? "RTP" : "RTCP") << " packet");
+  if (is_rtp) {
+    pipeline_->increment_rtp_packets_sent(out_len);
+  } else {
+    pipeline_->increment_rtcp_packets_sent();
+  }
+  return pipeline_->SendPacket(transport.transport_, data->data(), out_len);
+}
+
+nsresult MediaPipeline::PipelineTransport::SendRtcpPacket(
+    const void *data, int len) {
+
+    nsAutoPtr<DataBuffer> buf(new DataBuffer(static_cast<const uint8_t *>(data),
+                                             len, len + SRTP_MAX_EXPANSION));
+
+    RUN_ON_THREAD(sts_thread_,
+                  WrapRunnable(
+                      RefPtr<MediaPipeline::PipelineTransport>(this),
+                      &MediaPipeline::PipelineTransport::SendRtpRtcpPacket_s,
+                      buf, false),
+                  NS_DISPATCH_NORMAL);
+
+    return NS_OK;
+}
+
+void MediaPipelineTransmit::PipelineListener::
+UnsetTrackId(MediaStreamGraphImpl* graph) {
+#ifndef USE_FAKE_MEDIA_STREAMS
+  class Message : public ControlMessage {
+  public:
+    explicit Message(PipelineListener* listener) :
+      ControlMessage(nullptr), listener_(listener) {}
+    virtual void Run() override
+    {
+      listener_->UnsetTrackIdImpl();
+    }
+    RefPtr<PipelineListener> listener_;
+  };
+  graph->AppendMessage(MakeUnique<Message>(this));
+#else
+  UnsetTrackIdImpl();
+#endif
+}
+// Called if we're attached with AddDirectListener()
+void MediaPipelineTransmit::PipelineListener::
+NotifyRealtimeTrackData(MediaStreamGraph* graph,
+                        StreamTime offset,
+                        const MediaSegment& media) {
+  MOZ_MTLOG(ML_DEBUG, "MediaPipeline::NotifyRealtimeTrackData() listener=" <<
+                      this << ", offset=" << offset <<
+                      ", duration=" << media.GetDuration());
+
+  NewData(graph, offset, media);
+}
+
+void MediaPipelineTransmit::PipelineListener::
+NotifyQueuedChanges(MediaStreamGraph* graph,
+                    StreamTime offset,
+                    const MediaSegment& queued_media) {
+  MOZ_MTLOG(ML_DEBUG, "MediaPipeline::NotifyQueuedChanges()");
+
+  // ignore non-direct data if we're also getting direct data
+  if (!direct_connect_) {
+    NewData(graph, offset, queued_media);
+  }
+}
+
+void MediaPipelineTransmit::PipelineListener::
+NotifyDirectListenerInstalled(InstallationResult aResult) {
+  MOZ_MTLOG(ML_INFO, "MediaPipeline::NotifyDirectListenerInstalled() listener= " <<
+                     this << ", result=" << static_cast<int32_t>(aResult));
+
+  direct_connect_ = InstallationResult::SUCCESS == aResult;
+}
+
+void MediaPipelineTransmit::PipelineListener::
+NotifyDirectListenerUninstalled() {
+  MOZ_MTLOG(ML_INFO, "MediaPipeline::NotifyDirectListenerUninstalled() listener=" << this);
+
+  direct_connect_ = false;
+}
+
+void MediaPipelineTransmit::PipelineListener::
+NewData(MediaStreamGraph* graph,
+        StreamTime offset,
+        const MediaSegment& media) {
+  if (!active_) {
+    MOZ_MTLOG(ML_DEBUG, "Discarding packets because transport not ready");
+    return;
+  }
+
+  if (conduit_->type() !=
+      (media.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 (media.GetType() == MediaSegment::AUDIO) {
+    AudioSegment* audio = const_cast<AudioSegment *>(
+        static_cast<const AudioSegment *>(&media));
+
+    AudioSegment::ChunkIterator iter(*audio);
+    while(!iter.IsEnded()) {
+      TrackRate rate;
+#ifdef USE_FAKE_MEDIA_STREAMS
+      rate = Fake_MediaStream::GraphRate();
+#else
+      rate = graph->GraphRate();
+#endif
+      audio_processing_->QueueAudioChunk(rate, *iter, enabled_);
+      iter.Next();
+    }
+  } else {
+    // Ignore
+  }
+}
+
+void MediaPipelineTransmit::PipelineVideoSink::
+SetCurrentFrames(const VideoSegment& aSegment)
+{
+  MOZ_ASSERT(pipelineListener_);
+
+  if (!pipelineListener_->active_) {
+    MOZ_MTLOG(ML_DEBUG, "Discarding packets because transport not ready");
+    return;
+  }
+
+  if (conduit_->type() != MediaSessionConduit::VIDEO) {
+    // Ignore data of wrong kind in case we have a muxed stream
+    return;
+  }
+
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+    VideoSegment* video = const_cast<VideoSegment *>(&aSegment);
+
+    VideoSegment::ChunkIterator iter(*video);
+    while(!iter.IsEnded()) {
+      pipelineListener_->converter_->QueueVideoChunk(*iter, !pipelineListener_->enabled_);
+      iter.Next();
+    }
+#endif
+}
+
+class TrackAddedCallback {
+ public:
+  virtual void TrackAdded(TrackTicks current_ticks) = 0;
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(TrackAddedCallback);
+
+ protected:
+  virtual ~TrackAddedCallback() {}
+};
+
+class GenericReceiveListener;
+
+class GenericReceiveCallback : public TrackAddedCallback
+{
+ public:
+  explicit GenericReceiveCallback(GenericReceiveListener* listener)
+    : listener_(listener) {}
+
+  void TrackAdded(TrackTicks time);
+
+ private:
+  RefPtr<GenericReceiveListener> listener_;
+};
+
+// Add a listener on the MSG thread using the MSG command queue
+static void AddListener(MediaStream* source, MediaStreamListener* listener) {
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  class Message : public ControlMessage {
+   public:
+    Message(MediaStream* stream, MediaStreamListener* listener)
+      : ControlMessage(stream),
+        listener_(listener) {}
+
+    virtual void Run() override {
+      mStream->AddListenerImpl(listener_.forget());
+    }
+   private:
+    RefPtr<MediaStreamListener> listener_;
+  };
+
+  MOZ_ASSERT(listener);
+
+  source->GraphImpl()->AppendMessage(MakeUnique<Message>(source, listener));
+#else
+  source->AddListener(listener);
+#endif
+}
+
+class GenericReceiveListener : public MediaStreamListener
+{
+ public:
+  GenericReceiveListener(SourceMediaStream *source, TrackID track_id)
+    : source_(source),
+      track_id_(track_id),
+      played_ticks_(0),
+      principal_handle_(PRINCIPAL_HANDLE_NONE) {}
+
+  virtual ~GenericReceiveListener() {}
+
+  void AddSelf()
+  {
+    AddListener(source_, this);
+  }
+
+  void EndTrack()
+  {
+    source_->EndTrack(track_id_);
+  }
+
+#ifndef USE_FAKE_MEDIA_STREAMS
+  // Must be called on the main thread
+  void SetPrincipalHandle_m(const PrincipalHandle& principal_handle)
+  {
+    class Message : public ControlMessage
+    {
+    public:
+      Message(GenericReceiveListener* listener,
+              MediaStream* stream,
+              const PrincipalHandle& principal_handle)
+        : ControlMessage(stream),
+          listener_(listener),
+          principal_handle_(principal_handle)
+      {}
+
+      void Run() override {
+        listener_->SetPrincipalHandle_msg(principal_handle_);
+      }
+
+      RefPtr<GenericReceiveListener> listener_;
+      PrincipalHandle principal_handle_;
+    };
+
+    source_->GraphImpl()->AppendMessage(MakeUnique<Message>(this, source_, principal_handle));
+  }
+
+  // Must be called on the MediaStreamGraph thread
+  void SetPrincipalHandle_msg(const PrincipalHandle& principal_handle)
+  {
+    principal_handle_ = principal_handle;
+  }
+#endif // USE_FAKE_MEDIA_STREAMS
+
+ protected:
+  SourceMediaStream *source_;
+  const TrackID track_id_;
+  TrackTicks played_ticks_;
+  PrincipalHandle principal_handle_;
+};
+
+MediaPipelineReceive::MediaPipelineReceive(
+    const std::string& pc,
+    nsCOMPtr<nsIEventTarget> main_thread,
+    nsCOMPtr<nsIEventTarget> sts_thread,
+    SourceMediaStream *stream,
+    const std::string& track_id,
+    int level,
+    RefPtr<MediaSessionConduit> conduit,
+    RefPtr<TransportFlow> rtp_transport,
+    RefPtr<TransportFlow> rtcp_transport,
+    nsAutoPtr<MediaPipelineFilter> filter) :
+  MediaPipeline(pc, RECEIVE, main_thread, sts_thread,
+                track_id, level, conduit, rtp_transport,
+                rtcp_transport, filter),
+  stream_(stream),
+  segments_added_(0)
+{
+  MOZ_ASSERT(stream_);
+}
+
+MediaPipelineReceive::~MediaPipelineReceive()
+{
+  MOZ_ASSERT(!stream_);  // Check that we have shut down already.
+}
+
+class MediaPipelineReceiveAudio::PipelineListener
+  : public GenericReceiveListener
+{
+public:
+  PipelineListener(SourceMediaStream * source, TrackID track_id,
+                   const RefPtr<MediaSessionConduit>& conduit)
+    : GenericReceiveListener(source, track_id),
+      conduit_(conduit)
+  {
+  }
+
+  ~PipelineListener()
+  {
+    if (!NS_IsMainThread()) {
+      // release conduit on mainthread.  Must use forget()!
+      nsresult rv = NS_DispatchToMainThread(new
+                                            ConduitDeleteEvent(conduit_.forget()));
+      MOZ_ASSERT(!NS_FAILED(rv),"Could not dispatch conduit shutdown to main");
+      if (NS_FAILED(rv)) {
+        MOZ_CRASH();
+      }
+    } else {
+      conduit_ = nullptr;
+    }
+  }
+
+  // Implement MediaStreamListener
+  void NotifyPull(MediaStreamGraph* graph, StreamTime desired_time) override
+  {
+    MOZ_ASSERT(source_);
+    if (!source_) {
+      MOZ_MTLOG(ML_ERROR, "NotifyPull() called from a non-SourceMediaStream");
+      return;
+    }
+
+    // This comparison is done in total time to avoid accumulated roundoff errors.
+    while (source_->TicksToTimeRoundDown(WEBRTC_DEFAULT_SAMPLE_RATE,
+                                         played_ticks_) < desired_time) {
+      int16_t scratch_buffer[AUDIO_SAMPLE_BUFFER_MAX];
+
+      int samples_length;
+
+      // This fetches 10ms of data, either mono or stereo
+      MediaConduitErrorCode err =
+          static_cast<AudioSessionConduit*>(conduit_.get())->GetAudioFrame(
+              scratch_buffer,
+              WEBRTC_DEFAULT_SAMPLE_RATE,
+              0,  // TODO(ekr@rtfm.com): better estimate of "capture" (really playout) delay
+              samples_length);
+
+      if (err != kMediaConduitNoError) {
+        // Insert silence on conduit/GIPS failure (extremely unlikely)
+        MOZ_MTLOG(ML_ERROR, "Audio conduit failed (" << err
+                  << ") to return data @ " << played_ticks_
+                  << " (desired " << desired_time << " -> "
+                  << source_->StreamTimeToSeconds(desired_time) << ")");
+        // if this is not enough we'll loop and provide more
+        samples_length = WEBRTC_DEFAULT_SAMPLE_RATE/100;
+        PodArrayZero(scratch_buffer);
+      }
+
+      MOZ_ASSERT(samples_length * sizeof(uint16_t) < AUDIO_SAMPLE_BUFFER_MAX);
+
+      MOZ_MTLOG(ML_DEBUG, "Audio conduit returned buffer of length "
+                << samples_length);
+
+      RefPtr<SharedBuffer> samples = SharedBuffer::Create(samples_length * sizeof(uint16_t));
+      int16_t *samples_data = static_cast<int16_t *>(samples->Data());
+      AudioSegment segment;
+      // We derive the number of channels of the stream from the number of samples
+      // the AudioConduit gives us, considering it gives us packets of 10ms and we
+      // know the rate.
+      uint32_t channelCount = samples_length / (WEBRTC_DEFAULT_SAMPLE_RATE / 100);
+      AutoTArray<int16_t*,2> channels;
+      AutoTArray<const int16_t*,2> outputChannels;
+      size_t frames = samples_length / channelCount;
+
+      channels.SetLength(channelCount);
+
+      size_t offset = 0;
+      for (size_t i = 0; i < channelCount; i++) {
+        channels[i] = samples_data + offset;
+        offset += frames;
+      }
+
+      DeinterleaveAndConvertBuffer(scratch_buffer,
+                                   frames,
+                                   channelCount,
+                                   channels.Elements());
+
+      outputChannels.AppendElements(channels);
+
+      segment.AppendFrames(samples.forget(), outputChannels, frames,
+                           principal_handle_);
+
+      // Handle track not actually added yet or removed/finished
+      if (source_->AppendToTrack(track_id_, &segment)) {
+        played_ticks_ += frames;
+      } else {
+        MOZ_MTLOG(ML_ERROR, "AppendToTrack failed");
+        // we can't un-read the data, but that's ok since we don't want to
+        // buffer - but don't i-loop!
+        return;
+      }
+    }
+  }
+
+private:
+  RefPtr<MediaSessionConduit> conduit_;
+};
+
+MediaPipelineReceiveAudio::MediaPipelineReceiveAudio(
+    const std::string& pc,
+    nsCOMPtr<nsIEventTarget> main_thread,
+    nsCOMPtr<nsIEventTarget> sts_thread,
+    SourceMediaStream* stream,
+    const std::string& media_stream_track_id,
+    TrackID numeric_track_id,
+    int level,
+    RefPtr<AudioSessionConduit> conduit,
+    RefPtr<TransportFlow> rtp_transport,
+    RefPtr<TransportFlow> rtcp_transport,
+    nsAutoPtr<MediaPipelineFilter> filter) :
+  MediaPipelineReceive(pc, main_thread, sts_thread,
+                       stream, media_stream_track_id, level, conduit,
+                       rtp_transport, rtcp_transport, filter),
+  listener_(new PipelineListener(stream, numeric_track_id, conduit))
+{}
+
+void MediaPipelineReceiveAudio::DetachMedia()
+{
+  ASSERT_ON_THREAD(main_thread_);
+  if (stream_ && listener_) {
+    listener_->EndTrack();
+    stream_->RemoveListener(listener_);
+    stream_ = nullptr;
+  }
+}
+
+nsresult MediaPipelineReceiveAudio::Init() {
+  ASSERT_ON_THREAD(main_thread_);
+  MOZ_MTLOG(ML_DEBUG, __FUNCTION__);
+
+  description_ = pc_ + "| Receive audio[";
+  description_ += track_id_;
+  description_ += "]";
+
+  listener_->AddSelf();
+
+  return MediaPipelineReceive::Init();
+}
+
+#ifndef USE_FAKE_MEDIA_STREAMS
+void MediaPipelineReceiveAudio::SetPrincipalHandle_m(const PrincipalHandle& principal_handle)
+{
+  listener_->SetPrincipalHandle_m(principal_handle);
+}
+#endif // USE_FAKE_MEDIA_STREAMS
+
+class MediaPipelineReceiveVideo::PipelineListener
+  : public GenericReceiveListener {
+public:
+  PipelineListener(SourceMediaStream * source, TrackID track_id)
+    : GenericReceiveListener(source, track_id),
+      width_(0),
+      height_(0),
+#if defined(MOZILLA_INTERNAL_API)
+      image_container_(),
+      image_(),
+#endif
+      monitor_("Video PipelineListener")
+  {
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+    image_container_ =
+      LayerManager::CreateImageContainer(ImageContainer::ASYNCHRONOUS);
+#endif
+  }
+
+  // Implement MediaStreamListener
+  void NotifyPull(MediaStreamGraph* graph, StreamTime desired_time) override
+  {
+  #if defined(MOZILLA_INTERNAL_API)
+    ReentrantMonitorAutoEnter enter(monitor_);
+
+    RefPtr<Image> image = image_;
+    StreamTime delta = desired_time - played_ticks_;
+
+    // Don't append if we've already provided a frame that supposedly
+    // goes past the current aDesiredTime Doing so means a negative
+    // delta and thus messes up handling of the graph
+    if (delta > 0) {
+      VideoSegment segment;
+      segment.AppendFrame(image.forget(), delta, IntSize(width_, height_),
+                          principal_handle_);
+      // Handle track not actually added yet or removed/finished
+      if (source_->AppendToTrack(track_id_, &segment)) {
+        played_ticks_ = desired_time;
+      } else {
+        MOZ_MTLOG(ML_ERROR, "AppendToTrack failed");
+        return;
+      }
+    }
+  #endif
+  }
+
+  // Accessors for external writes from the renderer
+  void FrameSizeChange(unsigned int width,
+                       unsigned int height,
+                       unsigned int number_of_streams) {
+    ReentrantMonitorAutoEnter enter(monitor_);
+
+    width_ = width;
+    height_ = height;
+  }
+
+  void RenderVideoFrame(const unsigned char* buffer,
+                        size_t buffer_size,
+                        uint32_t time_stamp,
+                        int64_t render_time,
+                        const RefPtr<layers::Image>& video_image)
+  {
+    RenderVideoFrame(buffer, buffer_size, width_, (width_ + 1) >> 1,
+                     time_stamp, render_time, video_image);
+  }
+
+  void RenderVideoFrame(const unsigned char* buffer,
+                        size_t buffer_size,
+                        uint32_t y_stride,
+                        uint32_t cbcr_stride,
+                        uint32_t time_stamp,
+                        int64_t render_time,
+                        const RefPtr<layers::Image>& video_image)
+  {
+#ifdef MOZILLA_INTERNAL_API
+    ReentrantMonitorAutoEnter enter(monitor_);
+#endif // MOZILLA_INTERNAL_API
+
+#if defined(MOZILLA_INTERNAL_API)
+    if (buffer) {
+      // Create a video frame using |buffer|.
+#ifdef MOZ_WIDGET_GONK
+      RefPtr<PlanarYCbCrImage> yuvImage = new GrallocImage();
+#else
+      RefPtr<PlanarYCbCrImage> yuvImage = image_container_->CreatePlanarYCbCrImage();
+#endif
+      uint8_t* frame = const_cast<uint8_t*>(static_cast<const uint8_t*> (buffer));
+
+      PlanarYCbCrData yuvData;
+      yuvData.mYChannel = frame;
+      yuvData.mYSize = IntSize(y_stride, height_);
+      yuvData.mYStride = y_stride;
+      yuvData.mCbCrStride = cbcr_stride;
+      yuvData.mCbChannel = frame + height_ * yuvData.mYStride;
+      yuvData.mCrChannel = yuvData.mCbChannel + ((height_ + 1) >> 1) * yuvData.mCbCrStride;
+      yuvData.mCbCrSize = IntSize(yuvData.mCbCrStride, (height_ + 1) >> 1);
+      yuvData.mPicX = 0;
+      yuvData.mPicY = 0;
+      yuvData.mPicSize = IntSize(width_, height_);
+      yuvData.mStereoMode = StereoMode::MONO;
+
+      if (!yuvImage->CopyData(yuvData)) {
+        MOZ_ASSERT(false);
+        return;
+      }
+
+      image_ = yuvImage;
+    }
+#ifdef WEBRTC_GONK
+    else {
+      // Decoder produced video frame that can be appended to the track directly.
+      MOZ_ASSERT(video_image);
+      image_ = video_image;
+    }
+#endif // WEBRTC_GONK
+#endif // MOZILLA_INTERNAL_API
+  }
+
+private:
+  int width_;
+  int height_;
+#if defined(MOZILLA_INTERNAL_API)
+  RefPtr<layers::ImageContainer> image_container_;
+  RefPtr<layers::Image> image_;
+#endif
+  mozilla::ReentrantMonitor monitor_; // Monitor for processing WebRTC frames.
+                                      // Protects image_ against:
+                                      // - Writing from the GIPS thread
+                                      // - Reading from the MSG thread
+};
+
+class MediaPipelineReceiveVideo::PipelineRenderer : public VideoRenderer
+{
+public:
+  explicit PipelineRenderer(MediaPipelineReceiveVideo *pipeline) :
+    pipeline_(pipeline) {}
+
+  void Detach() { pipeline_ = nullptr; }
+
+  // Implement VideoRenderer
+  void FrameSizeChange(unsigned int width,
+                       unsigned int height,
+                       unsigned int number_of_streams) override
+  {
+    pipeline_->listener_->FrameSizeChange(width, height, number_of_streams);
+  }
+
+  void RenderVideoFrame(const unsigned char* buffer,
+                        size_t buffer_size,
+                        uint32_t time_stamp,
+                        int64_t render_time,
+                        const ImageHandle& handle) override
+  {
+    pipeline_->listener_->RenderVideoFrame(buffer, buffer_size,
+                                           time_stamp, render_time,
+                                           handle.GetImage());
+  }
+
+  void RenderVideoFrame(const unsigned char* buffer,
+                        size_t buffer_size,
+                        uint32_t y_stride,
+                        uint32_t cbcr_stride,
+                        uint32_t time_stamp,
+                        int64_t render_time,
+                        const ImageHandle& handle) override
+  {
+    pipeline_->listener_->RenderVideoFrame(buffer, buffer_size,
+                                           y_stride, cbcr_stride,
+                                           time_stamp, render_time,
+                                           handle.GetImage());
+  }
+
+private:
+  MediaPipelineReceiveVideo *pipeline_;  // Raw pointer to avoid cycles
+};
+
+
+MediaPipelineReceiveVideo::MediaPipelineReceiveVideo(
+    const std::string& pc,
+    nsCOMPtr<nsIEventTarget> main_thread,
+    nsCOMPtr<nsIEventTarget> sts_thread,
+    SourceMediaStream *stream,
+    const std::string& media_stream_track_id,
+    TrackID numeric_track_id,
+    int level,
+    RefPtr<VideoSessionConduit> conduit,
+    RefPtr<TransportFlow> rtp_transport,
+    RefPtr<TransportFlow> rtcp_transport,
+    nsAutoPtr<MediaPipelineFilter> filter) :
+  MediaPipelineReceive(pc, main_thread, sts_thread,
+                       stream, media_stream_track_id, level, conduit,
+                       rtp_transport, rtcp_transport, filter),
+  renderer_(new PipelineRenderer(this)),
+  listener_(new PipelineListener(stream, numeric_track_id))
+{}
+
+void MediaPipelineReceiveVideo::DetachMedia()
+{
+  ASSERT_ON_THREAD(main_thread_);
+
+  // 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*>(conduit_.get())->DetachRenderer();
+  if (stream_ && listener_) {
+    listener_->EndTrack();
+    stream_->RemoveListener(listener_);
+    stream_ = nullptr;
+  }
+}
+
+nsresult MediaPipelineReceiveVideo::Init() {
+  ASSERT_ON_THREAD(main_thread_);
+  MOZ_MTLOG(ML_DEBUG, __FUNCTION__);
+
+  description_ = pc_ + "| Receive video[";
+  description_ += track_id_;
+  description_ += "]";
+
+#if defined(MOZILLA_INTERNAL_API)
+  listener_->AddSelf();
+#endif
+
+  // Always happens before we can DetachMedia()
+  static_cast<VideoSessionConduit *>(conduit_.get())->
+      AttachRenderer(renderer_);
+
+  return MediaPipelineReceive::Init();
+}
+
+#ifndef USE_FAKE_MEDIA_STREAMS
+void MediaPipelineReceiveVideo::SetPrincipalHandle_m(const PrincipalHandle& principal_handle)
+{
+  listener_->SetPrincipalHandle_m(principal_handle);
+}
+#endif // USE_FAKE_MEDIA_STREAMS
+
+}  // end namespace
diff --git a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.h b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.h
new file mode 100644
index 000000000..d609cbd47
--- /dev/null
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.h
@@ -0,0 +1,479 @@
+/* -*- 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 "sigslot.h"
+
+#ifdef USE_FAKE_MEDIA_STREAMS
+#include "FakeMediaStreams.h"
+#endif
+#include "MediaConduitInterface.h"
+#include "mozilla/ReentrantMonitor.h"
+#include "mozilla/Atomics.h"
+#include "SrtpFlow.h"
+#include "databuffer.h"
+#include "runnable_utils.h"
+#include "transportflow.h"
+#include "AudioPacketizer.h"
+#include "StreamTracks.h"
+
+#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+
+// Should come from MediaEngine.h, but that's a pain to include here
+// because of the MOZILLA_EXTERNAL_LINKAGE stuff.
+#define WEBRTC_DEFAULT_SAMPLE_RATE 32000
+
+class nsIPrincipal;
+
+namespace mozilla {
+class MediaPipelineFilter;
+class PeerIdentity;
+class AudioProxyThread;
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+class VideoFrameConverter;
+#endif
+
+#ifndef USE_FAKE_MEDIA_STREAMS
+namespace dom {
+  class MediaStreamTrack;
+} // namespace dom
+
+class SourceMediaStream;
+#endif // USE_FAKE_MEDIA_STREAMS
+
+// 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
+// MediaStreamGraph
+//   * Receives outgoing data from the MediaStreamGraph
+//   * Receives pull requests for more data from the
+//     MediaStreamGraph
+// 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 Direction { TRANSMIT, RECEIVE };
+  enum State { MP_CONNECTING, MP_OPEN, MP_CLOSED };
+  MediaPipeline(const std::string& pc,
+                Direction direction,
+                nsCOMPtr<nsIEventTarget> main_thread,
+                nsCOMPtr<nsIEventTarget> sts_thread,
+                const std::string& track_id,
+                int level,
+                RefPtr<MediaSessionConduit> conduit,
+                RefPtr<TransportFlow> rtp_transport,
+                RefPtr<TransportFlow> rtcp_transport,
+                nsAutoPtr<MediaPipelineFilter> filter);
+
+  // Must be called on the STS thread.  Must be called after ShutdownMedia_m().
+  void DetachTransport_s();
+
+  // Must be called on the main thread.
+  void ShutdownMedia_m()
+  {
+    ASSERT_ON_THREAD(main_thread_);
+
+    if (direction_ == RECEIVE) {
+      conduit_->StopReceiving();
+    } else {
+      conduit_->StopTransmitting();
+    }
+    DetachMedia();
+  }
+
+  virtual nsresult Init();
+
+  void UpdateTransport_m(int level,
+                         RefPtr<TransportFlow> rtp_transport,
+                         RefPtr<TransportFlow> rtcp_transport,
+                         nsAutoPtr<MediaPipelineFilter> filter);
+
+  void UpdateTransport_s(int level,
+                         RefPtr<TransportFlow> rtp_transport,
+                         RefPtr<TransportFlow> rtcp_transport,
+                         nsAutoPtr<MediaPipelineFilter> filter);
+
+  // Used only for testing; installs a MediaPipelineFilter that filters
+  // everything but the nth ssrc
+  void SelectSsrc_m(size_t ssrc_index);
+  void SelectSsrc_s(size_t ssrc_index);
+
+  virtual Direction direction() const { return direction_; }
+  virtual const std::string& trackid() const { return track_id_; }
+  virtual int level() const { return level_; }
+  virtual bool IsVideo() const = 0;
+
+  bool IsDoingRtcpMux() const {
+    return (rtp_.type_ == MUX);
+  }
+
+  int32_t rtp_packets_sent() const { return rtp_packets_sent_; }
+  int64_t rtp_bytes_sent() const { return rtp_bytes_sent_; }
+  int32_t rtcp_packets_sent() const { return rtcp_packets_sent_; }
+  int32_t rtp_packets_received() const { return rtp_packets_received_; }
+  int64_t rtp_bytes_received() const { return rtp_bytes_received_; }
+  int32_t rtcp_packets_received() const { return rtcp_packets_received_; }
+
+  MediaSessionConduit *Conduit() const { return conduit_; }
+
+  // Thread counting
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaPipeline)
+
+  typedef enum {
+    RTP,
+    RTCP,
+    MUX,
+    MAX_RTP_TYPE
+  } RtpType;
+
+ protected:
+  virtual ~MediaPipeline();
+  virtual void DetachMedia() {}
+  nsresult AttachTransport_s();
+
+  // Separate class to allow ref counting
+  class PipelineTransport : public TransportInterface {
+   public:
+    // Implement the TransportInterface functions
+    explicit PipelineTransport(MediaPipeline *pipeline)
+        : pipeline_(pipeline),
+          sts_thread_(pipeline->sts_thread_) {}
+
+    void Attach(MediaPipeline *pipeline) { pipeline_ = pipeline; }
+    void Detach() { pipeline_ = nullptr; }
+    MediaPipeline *pipeline() const { return pipeline_; }
+
+    virtual nsresult SendRtpPacket(const void* data, int len);
+    virtual nsresult SendRtcpPacket(const void* data, int len);
+
+   private:
+    nsresult SendRtpRtcpPacket_s(nsAutoPtr<DataBuffer> data,
+                                 bool is_rtp);
+
+    MediaPipeline *pipeline_;  // Raw pointer to avoid cycles
+    nsCOMPtr<nsIEventTarget> sts_thread_;
+  };
+  friend class PipelineTransport;
+
+  class TransportInfo {
+    public:
+      TransportInfo(RefPtr<TransportFlow> flow, RtpType type) :
+        transport_(flow),
+        state_(MP_CONNECTING),
+        type_(type) {
+        MOZ_ASSERT(flow);
+      }
+
+      void Detach()
+      {
+        transport_ = nullptr;
+        send_srtp_ = nullptr;
+        recv_srtp_ = nullptr;
+      }
+
+      RefPtr<TransportFlow> transport_;
+      State state_;
+      RefPtr<SrtpFlow> send_srtp_;
+      RefPtr<SrtpFlow> recv_srtp_;
+      RtpType type_;
+  };
+
+  // The transport is down
+  virtual nsresult TransportFailed_s(TransportInfo &info);
+  // The transport is ready
+  virtual nsresult TransportReady_s(TransportInfo &info);
+  void UpdateRtcpMuxState(TransportInfo &info);
+
+  // Unhooks from signals
+  void DisconnectTransport_s(TransportInfo &info);
+  nsresult ConnectTransport_s(TransportInfo &info);
+
+  TransportInfo* GetTransportInfo_s(TransportFlow *flow);
+
+  void increment_rtp_packets_sent(int bytes);
+  void increment_rtcp_packets_sent();
+  void increment_rtp_packets_received(int bytes);
+  void increment_rtcp_packets_received();
+
+  virtual nsresult SendPacket(TransportFlow *flow, const void *data, int len);
+
+  // Process slots on transports
+  void StateChange(TransportFlow *flow, TransportLayer::State);
+  void RtpPacketReceived(TransportLayer *layer, const unsigned char *data,
+                         size_t len);
+  void RtcpPacketReceived(TransportLayer *layer, const unsigned char *data,
+                          size_t len);
+  void PacketReceived(TransportLayer *layer, const unsigned char *data,
+                      size_t len);
+
+  Direction direction_;
+  std::string track_id_;        // The track on the stream.
+                                // Written on the main thread.
+                                // Used on STS and MediaStreamGraph threads.
+                                // Not used outside initialization in MediaPipelineTransmit
+  // The m-line index (starting at 0, to match convention) Atomic because
+  // this value is updated from STS, but read on main, and we don't want to
+  // bother with dispatches just to get an int occasionally.
+  Atomic<int> level_;
+  RefPtr<MediaSessionConduit> conduit_;  // Our conduit. Written on the main
+                                         // thread. Read on STS thread.
+
+  // The transport objects. Read/written on STS thread.
+  TransportInfo rtp_;
+  TransportInfo rtcp_;
+
+  // Pointers to the threads we need. Initialized at creation
+  // and used all over the place.
+  nsCOMPtr<nsIEventTarget> main_thread_;
+  nsCOMPtr<nsIEventTarget> sts_thread_;
+
+  // Created on Init. Referenced by the conduit and eventually
+  // destroyed on the STS thread.
+  RefPtr<PipelineTransport> transport_;
+
+  // Only safe to access from STS thread.
+  // Build into TransportInfo?
+  int32_t rtp_packets_sent_;
+  int32_t rtcp_packets_sent_;
+  int32_t rtp_packets_received_;
+  int32_t rtcp_packets_received_;
+  int64_t rtp_bytes_sent_;
+  int64_t rtp_bytes_received_;
+
+  std::vector<uint32_t> ssrcs_received_;
+
+  // Written on Init. Read on STS thread.
+  std::string pc_;
+  std::string description_;
+
+  // Written on Init, all following accesses are on the STS thread.
+  nsAutoPtr<MediaPipelineFilter> filter_;
+  nsAutoPtr<webrtc::RtpHeaderParser> rtp_parser_;
+
+ private:
+  nsresult Init_s();
+
+  bool IsRtp(const unsigned char *data, size_t len);
+};
+
+class ConduitDeleteEvent: public Runnable
+{
+public:
+  explicit ConduitDeleteEvent(already_AddRefed<MediaSessionConduit> aConduit) :
+    mConduit(aConduit) {}
+
+  /* we exist solely to proxy release of the conduit */
+  NS_IMETHOD Run() override { return NS_OK; }
+private:
+  RefPtr<MediaSessionConduit> mConduit;
+};
+
+// A specialization of pipeline for reading from an input device
+// and transmitting to the network.
+class MediaPipelineTransmit : public MediaPipeline {
+public:
+  // Set rtcp_transport to nullptr to use rtcp-mux
+  MediaPipelineTransmit(const std::string& pc,
+                        nsCOMPtr<nsIEventTarget> main_thread,
+                        nsCOMPtr<nsIEventTarget> sts_thread,
+                        dom::MediaStreamTrack* domtrack,
+                        const std::string& track_id,
+                        int level,
+                        RefPtr<MediaSessionConduit> conduit,
+                        RefPtr<TransportFlow> rtp_transport,
+                        RefPtr<TransportFlow> rtcp_transport,
+                        nsAutoPtr<MediaPipelineFilter> filter);
+
+  // Initialize (stuff here may fail)
+  nsresult Init() override;
+
+  virtual void AttachToTrack(const std::string& track_id);
+
+  // written and used from MainThread
+  bool IsVideo() const override;
+
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  // When the principal of the domtrack changes, it calls through to here
+  // so that we can determine whether to enable track transmission.
+  // `track` has to be null or equal `domtrack_` for us to apply the update.
+  virtual void UpdateSinkIdentity_m(dom::MediaStreamTrack* track,
+                                    nsIPrincipal* principal,
+                                    const PeerIdentity* sinkIdentity);
+#endif
+
+  // Called on the main thread.
+  void DetachMedia() override;
+
+  // Override MediaPipeline::TransportReady.
+  nsresult TransportReady_s(TransportInfo &info) override;
+
+  // Replace a track with a different one
+  // In non-compliance with the likely final spec, allow the new
+  // track to be part of a different stream (since we don't support
+  // multiple tracks of a type in a stream yet).  bug 1056650
+  virtual nsresult ReplaceTrack(dom::MediaStreamTrack& domtrack);
+
+  // Separate classes to allow ref counting
+  class PipelineListener;
+  class VideoFrameFeeder;
+  class PipelineVideoSink;
+
+ protected:
+  ~MediaPipelineTransmit();
+
+ private:
+  RefPtr<PipelineListener> listener_;
+  RefPtr<AudioProxyThread> audio_processing_;
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  RefPtr<VideoFrameFeeder> feeder_;
+  RefPtr<VideoFrameConverter> converter_;
+#endif
+  RefPtr<PipelineVideoSink> video_sink_;
+  dom::MediaStreamTrack* domtrack_;
+};
+
+
+// A specialization of pipeline for reading from the network and
+// rendering video.
+class MediaPipelineReceive : public MediaPipeline {
+ public:
+  // Set rtcp_transport to nullptr to use rtcp-mux
+  MediaPipelineReceive(const std::string& pc,
+                       nsCOMPtr<nsIEventTarget> main_thread,
+                       nsCOMPtr<nsIEventTarget> sts_thread,
+                       SourceMediaStream *stream,
+                       const std::string& track_id,
+                       int level,
+                       RefPtr<MediaSessionConduit> conduit,
+                       RefPtr<TransportFlow> rtp_transport,
+                       RefPtr<TransportFlow> rtcp_transport,
+                       nsAutoPtr<MediaPipelineFilter> filter);
+
+  int segments_added() const { return segments_added_; }
+
+#ifndef USE_FAKE_MEDIA_STREAMS
+  // Sets the PrincipalHandle we set on the media chunks produced by this
+  // pipeline. Must be called on the main thread.
+  virtual void SetPrincipalHandle_m(const PrincipalHandle& principal_handle) = 0;
+#endif // USE_FAKE_MEDIA_STREAMS
+ protected:
+  ~MediaPipelineReceive();
+
+  RefPtr<SourceMediaStream> stream_;
+  int segments_added_;
+
+ private:
+};
+
+
+// A specialization of pipeline for reading from the network and
+// rendering audio.
+class MediaPipelineReceiveAudio : public MediaPipelineReceive {
+ public:
+  MediaPipelineReceiveAudio(const std::string& pc,
+                            nsCOMPtr<nsIEventTarget> main_thread,
+                            nsCOMPtr<nsIEventTarget> sts_thread,
+                            SourceMediaStream* stream,
+                            // This comes from an msid attribute. Everywhere
+                            // but MediaStreamGraph uses this.
+                            const std::string& media_stream_track_id,
+                            // This is an integer identifier that is only
+                            // unique within a single DOMMediaStream, which is
+                            // used by MediaStreamGraph
+                            TrackID numeric_track_id,
+                            int level,
+                            RefPtr<AudioSessionConduit> conduit,
+                            RefPtr<TransportFlow> rtp_transport,
+                            RefPtr<TransportFlow> rtcp_transport,
+                            nsAutoPtr<MediaPipelineFilter> filter);
+
+  void DetachMedia() override;
+
+  nsresult Init() override;
+  bool IsVideo() const override { return false; }
+
+#ifndef USE_FAKE_MEDIA_STREAMS
+  void SetPrincipalHandle_m(const PrincipalHandle& principal_handle) override;
+#endif // USE_FAKE_MEDIA_STREAMS
+
+ private:
+  // Separate class to allow ref counting
+  class PipelineListener;
+
+  RefPtr<PipelineListener> listener_;
+};
+
+
+// A specialization of pipeline for reading from the network and
+// rendering video.
+class MediaPipelineReceiveVideo : public MediaPipelineReceive {
+ public:
+  MediaPipelineReceiveVideo(const std::string& pc,
+                            nsCOMPtr<nsIEventTarget> main_thread,
+                            nsCOMPtr<nsIEventTarget> sts_thread,
+                            SourceMediaStream *stream,
+                            // This comes from an msid attribute. Everywhere
+                            // but MediaStreamGraph uses this.
+                            const std::string& media_stream_track_id,
+                            // This is an integer identifier that is only
+                            // unique within a single DOMMediaStream, which is
+                            // used by MediaStreamGraph
+                            TrackID numeric_track_id,
+                            int level,
+                            RefPtr<VideoSessionConduit> conduit,
+                            RefPtr<TransportFlow> rtp_transport,
+                            RefPtr<TransportFlow> rtcp_transport,
+                            nsAutoPtr<MediaPipelineFilter> filter);
+
+  // Called on the main thread.
+  void DetachMedia() override;
+
+  nsresult Init() override;
+  bool IsVideo() const override { return true; }
+
+#ifndef USE_FAKE_MEDIA_STREAMS
+  void SetPrincipalHandle_m(const PrincipalHandle& principal_handle) override;
+#endif // USE_FAKE_MEDIA_STREAMS
+
+ private:
+  class PipelineRenderer;
+  friend class PipelineRenderer;
+
+  // Separate class to allow ref counting
+  class PipelineListener;
+
+  RefPtr<PipelineRenderer> renderer_;
+  RefPtr<PipelineListener> listener_;
+};
+
+
+}  // namespace mozilla
+#endif
diff --git a/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp b/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp
new file mode 100644
index 000000000..b56c272f9
--- /dev/null
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp
@@ -0,0 +1,97 @@
+/* -*- 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 "webrtc/modules/interface/module_common_types.h"
+
+namespace mozilla {
+
+MediaPipelineFilter::MediaPipelineFilter() : correlator_(0) {
+}
+
+bool MediaPipelineFilter::Filter(const webrtc::RTPHeader& header,
+                                 uint32_t correlator) {
+  if (correlator) {
+    // This special correlator header takes precedence. It also lets us learn
+    // about SSRC mappings if we don't know about them yet.
+    if (correlator == correlator_) {
+      AddRemoteSSRC(header.ssrc);
+      return true;
+    } else {
+      // Some other stream; it is possible that an SSRC has moved, so make sure
+      // we don't have that SSRC in our filter any more.
+      remote_ssrc_set_.erase(header.ssrc);
+      return false;
+    }
+  }
+
+  if (remote_ssrc_set_.count(header.ssrc)) {
+    return true;
+  }
+
+  // Last ditch effort...
+  if (payload_type_set_.count(header.payloadType)) {
+    // 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;
+  }
+
+  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::SetCorrelator(uint32_t correlator) {
+  correlator_ = correlator;
+}
+
+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_;
+  }
+
+  payload_type_set_ = filter_update.payload_type_set_;
+  correlator_ = filter_update.correlator_;
+}
+
+bool
+MediaPipelineFilter::FilterSenderReport(const unsigned char* data,
+                                        size_t len) const {
+  if (len < FIRST_SSRC_OFFSET + 4) {
+    return false;
+  }
+
+  uint8_t payload_type = data[PT_OFFSET];
+
+  if (payload_type != SENDER_REPORT_T) {
+    return false;
+  }
+
+  uint32_t ssrc = 0;
+  ssrc += (uint32_t)data[FIRST_SSRC_OFFSET] << 24;
+  ssrc += (uint32_t)data[FIRST_SSRC_OFFSET + 1] << 16;
+  ssrc += (uint32_t)data[FIRST_SSRC_OFFSET + 2] << 8;
+  ssrc += (uint32_t)data[FIRST_SSRC_OFFSET + 3];
+
+  return !!remote_ssrc_set_.count(ssrc);
+}
+
+} // end namespace mozilla
+
diff --git a/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.h b/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.h
new file mode 100644
index 000000000..31de8ccb2
--- /dev/null
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.h
@@ -0,0 +1,86 @@
+/* -*- 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 <set>
+
+namespace webrtc {
+struct RTPHeader;
+}
+
+namespace mozilla {
+
+// 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();
+
+  // Checks whether this packet passes the filter, possibly updating the filter
+  // in the process (if the correlator or payload types are used, they can teach
+  // the filter about ssrcs)
+  bool Filter(const webrtc::RTPHeader& header, uint32_t correlator = 0);
+
+  // RTCP doesn't have things like the RTP correlator, and uses its own
+  // payload types too.
+  bool FilterSenderReport(const unsigned char* data, size_t len) const;
+
+  void AddRemoteSSRC(uint32_t ssrc);
+
+  // When a payload type id is unique to our media section, add it here.
+  void AddUniquePT(uint8_t payload_type);
+  void SetCorrelator(uint32_t correlator);
+
+  void Update(const MediaPipelineFilter& filter_update);
+
+  // Some payload types
+  static const uint8_t SENDER_REPORT_T = 200;
+
+ private:
+  // Payload type is always in the second byte
+  static const size_t PT_OFFSET = 1;
+  // First SSRC always starts at the fifth byte.
+  static const size_t FIRST_SSRC_OFFSET = 4;
+
+  uint32_t correlator_;
+  // 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_;
+};
+
+} // end namespace mozilla
+
+#endif // mediapipelinefilter_h__
+
diff --git a/media/webrtc/signaling/src/mediapipeline/SrtpFlow.cpp b/media/webrtc/signaling/src/mediapipeline/SrtpFlow.cpp
new file mode 100644
index 000000000..59c3dac0a
--- /dev/null
+++ b/media/webrtc/signaling/src/mediapipeline/SrtpFlow.cpp
@@ -0,0 +1,251 @@
+/* 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 "logging.h"
+#include "SrtpFlow.h"
+
+#include "srtp.h"
+#include "ssl.h"
+#include "sslproto.h"
+
+#include "mozilla/RefPtr.h"
+
+// Logging context
+using namespace mozilla;
+MOZ_MTLOG_MODULE("mediapipeline")
+
+namespace mozilla {
+
+bool SrtpFlow::initialized;  // Static
+
+SrtpFlow::~SrtpFlow() {
+  if (session_) {
+    srtp_dealloc(session_);
+  }
+}
+
+RefPtr<SrtpFlow> SrtpFlow::Create(int cipher_suite,
+                                           bool inbound,
+                                           const void *key,
+                                           size_t key_len) {
+  nsresult res = Init();
+  if (!NS_SUCCEEDED(res))
+    return nullptr;
+
+  RefPtr<SrtpFlow> flow = new SrtpFlow();
+
+  if (!key) {
+    MOZ_MTLOG(ML_ERROR, "Null SRTP key specified");
+    return nullptr;
+  }
+
+  if (key_len != SRTP_TOTAL_KEY_LENGTH) {
+    MOZ_MTLOG(ML_ERROR, "Invalid SRTP key length");
+    return nullptr;
+  }
+
+  srtp_policy_t policy;
+  memset(&policy, 0, sizeof(srtp_policy_t));
+
+  // Note that we set the same cipher suite for RTP and RTCP
+  // since any flow can only have one cipher suite with DTLS-SRTP
+  switch (cipher_suite) {
+    case SRTP_AES128_CM_HMAC_SHA1_80:
+      MOZ_MTLOG(ML_DEBUG,
+                "Setting SRTP cipher suite SRTP_AES128_CM_HMAC_SHA1_80");
+      crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
+      crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
+      break;
+    case SRTP_AES128_CM_HMAC_SHA1_32:
+      MOZ_MTLOG(ML_DEBUG,
+                "Setting SRTP cipher suite SRTP_AES128_CM_HMAC_SHA1_32");
+      crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp);
+      crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); // 80-bit per RFC 5764
+      break;                                                   // S 4.1.2.
+    default:
+      MOZ_MTLOG(ML_ERROR, "Request to set unknown SRTP cipher suite");
+      return nullptr;
+  }
+  // This key is copied into the srtp_t object, so we don't
+  // need to keep it.
+  policy.key = const_cast<unsigned char *>(
+      static_cast<const unsigned char *>(key));
+  policy.ssrc.type = inbound ? ssrc_any_inbound : ssrc_any_outbound;
+  policy.ssrc.value = 0;
+  policy.ekt = nullptr;
+  policy.window_size = 1024;   // Use the Chrome value.  Needs to be revisited.  Default is 128
+  policy.allow_repeat_tx = 1;  // Use Chrome value; needed for NACK mode to work
+  policy.next = nullptr;
+
+  // Now make the session
+  err_status_t r = srtp_create(&flow->session_, &policy);
+  if (r != err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error creating srtp session");
+    return nullptr;
+  }
+
+  return flow;
+}
+
+
+nsresult SrtpFlow::CheckInputs(bool protect, void *in, int in_len,
+                               int max_len, int *out_len) {
+  MOZ_ASSERT(in);
+  if (!in) {
+    MOZ_MTLOG(ML_ERROR, "NULL input value");
+    return NS_ERROR_NULL_POINTER;
+  }
+
+  if (in_len < 0) {
+    MOZ_MTLOG(ML_ERROR, "Input length is negative");
+    return NS_ERROR_ILLEGAL_VALUE;
+  }
+
+  if (max_len < 0) {
+    MOZ_MTLOG(ML_ERROR, "Max output length is negative");
+    return NS_ERROR_ILLEGAL_VALUE;
+  }
+
+  if (protect) {
+    if ((max_len < SRTP_MAX_EXPANSION) ||
+        ((max_len - SRTP_MAX_EXPANSION) < in_len)) {
+      MOZ_MTLOG(ML_ERROR, "Output too short");
+      return NS_ERROR_ILLEGAL_VALUE;
+    }
+  }
+  else {
+    if (in_len > max_len) {
+      MOZ_MTLOG(ML_ERROR, "Output too short");
+      return NS_ERROR_ILLEGAL_VALUE;
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult SrtpFlow::ProtectRtp(void *in, int in_len,
+                              int max_len, int *out_len) {
+  nsresult res = CheckInputs(true, in, in_len, max_len, out_len);
+  if (NS_FAILED(res))
+    return res;
+
+  int len = in_len;
+  err_status_t r = srtp_protect(session_, in, &len);
+
+  if (r != err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error protecting SRTP packet");
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_ASSERT(len <= max_len);
+  *out_len = len;
+
+
+  MOZ_MTLOG(ML_DEBUG, "Successfully protected an SRTP packet of len "
+            << *out_len);
+
+  return NS_OK;
+}
+
+nsresult SrtpFlow::UnprotectRtp(void *in, int in_len,
+                                int max_len, int *out_len) {
+  nsresult res = CheckInputs(false, in, in_len, max_len, out_len);
+  if (NS_FAILED(res))
+    return res;
+
+  int len = in_len;
+  err_status_t r = srtp_unprotect(session_, in, &len);
+
+  if (r != err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error unprotecting SRTP packet error=" << (int)r);
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_ASSERT(len <= max_len);
+  *out_len = len;
+
+  MOZ_MTLOG(ML_DEBUG, "Successfully unprotected an SRTP packet of len "
+            << *out_len);
+
+  return NS_OK;
+}
+
+nsresult SrtpFlow::ProtectRtcp(void *in, int in_len,
+                               int max_len, int *out_len) {
+  nsresult res = CheckInputs(true, in, in_len, max_len, out_len);
+  if (NS_FAILED(res))
+    return res;
+
+  int len = in_len;
+  err_status_t r = srtp_protect_rtcp(session_, in, &len);
+
+  if (r != err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error protecting SRTCP packet");
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_ASSERT(len <= max_len);
+  *out_len = len;
+
+  MOZ_MTLOG(ML_DEBUG, "Successfully protected an SRTCP packet of len "
+            << *out_len);
+
+  return NS_OK;
+}
+
+nsresult SrtpFlow::UnprotectRtcp(void *in, int in_len,
+                                 int max_len, int *out_len) {
+  nsresult res = CheckInputs(false, in, in_len, max_len, out_len);
+  if (NS_FAILED(res))
+    return res;
+
+  int len = in_len;
+  err_status_t r = srtp_unprotect_rtcp(session_, in, &len);
+
+  if (r != err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error unprotecting SRTCP packet error=" << (int)r);
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_ASSERT(len <= max_len);
+  *out_len = len;
+
+  MOZ_MTLOG(ML_DEBUG, "Successfully unprotected an SRTCP packet of len "
+            << *out_len);
+
+  return NS_OK;
+}
+
+// Statics
+void SrtpFlow::srtp_event_handler(srtp_event_data_t *data) {
+  // TODO(ekr@rtfm.com): Implement this
+  MOZ_CRASH();
+}
+
+nsresult SrtpFlow::Init() {
+  if (!initialized) {
+    err_status_t r = srtp_init();
+    if (r != err_status_ok) {
+      MOZ_MTLOG(ML_ERROR, "Could not initialize SRTP");
+      MOZ_ASSERT(PR_FALSE);
+      return NS_ERROR_FAILURE;
+    }
+
+    r = srtp_install_event_handler(&SrtpFlow::srtp_event_handler);
+    if (r != err_status_ok) {
+      MOZ_MTLOG(ML_ERROR, "Could not install SRTP event handler");
+      MOZ_ASSERT(PR_FALSE);
+      return NS_ERROR_FAILURE;
+    }
+
+    initialized = true;
+  }
+
+  return NS_OK;
+}
+
+}  // end of namespace
+
diff --git a/media/webrtc/signaling/src/mediapipeline/SrtpFlow.h b/media/webrtc/signaling/src/mediapipeline/SrtpFlow.h
new file mode 100644
index 000000000..9bb9c2a67
--- /dev/null
+++ b/media/webrtc/signaling/src/mediapipeline/SrtpFlow.h
@@ -0,0 +1,68 @@
+/* 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 srtpflow_h__
+#define srtpflow_h__
+
+#include "ssl.h"
+#include "sslproto.h"
+#include "mozilla/RefPtr.h"
+#include "nsISupportsImpl.h"
+
+typedef struct srtp_policy_t srtp_policy_t;
+typedef struct srtp_ctx_t *srtp_t;
+typedef struct srtp_event_data_t srtp_event_data_t;
+
+namespace mozilla {
+
+#define SRTP_MASTER_KEY_LENGTH 16
+#define SRTP_MASTER_SALT_LENGTH 14
+#define SRTP_TOTAL_KEY_LENGTH (SRTP_MASTER_KEY_LENGTH + SRTP_MASTER_SALT_LENGTH)
+
+// SRTCP requires an auth tag *plus* a 4-byte index-plus-'E'-bit value (see
+// RFC 3711)
+#define SRTP_MAX_EXPANSION (SRTP_MAX_TRAILER_LEN+4)
+
+
+class SrtpFlow {
+  ~SrtpFlow();
+ public:
+
+
+  static RefPtr<SrtpFlow> Create(int cipher_suite,
+                                          bool inbound,
+                                          const void *key,
+                                          size_t key_len);
+
+  nsresult ProtectRtp(void *in, int in_len,
+                      int max_len, int *out_len);
+  nsresult UnprotectRtp(void *in, int in_len,
+                        int max_len, int *out_len);
+  nsresult ProtectRtcp(void *in, int in_len,
+                       int max_len, int *out_len);
+  nsresult UnprotectRtcp(void *in, int in_len,
+                         int max_len, int *out_len);
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(SrtpFlow)
+
+  static void srtp_event_handler(srtp_event_data_t *data);
+
+
+ private:
+  SrtpFlow() : session_(nullptr) {}
+
+  nsresult CheckInputs(bool protect, void *in, int in_len,
+                       int max_len, int *out_len);
+
+  static nsresult Init();
+  static bool initialized;  // Was libsrtp initialized? Only happens once.
+
+  srtp_t session_;
+};
+
+}  // End of namespace
+#endif
+