Bug 1595479 - Add RtcpEventObserver to Conduits and Transceiver; r=ng

Differential Revision: https://phabricator.services.mozilla.com/D57860

--HG--
extra : moz-landing-system : lando

media/webrtc/signaling/src/media-conduit/AudioConduit.cpp

@@ -277,6 +277,34 @@ void WebrtcAudioConduit::OnRtpPacket(const webrtc::RTPHeader& aHeader,
   mRtpSourceObserver.OnRtpPacket(aHeader, aTimestamp, aJitter);
 }
 
+void WebrtcAudioConduit::OnRtcpBye() {
+  RefPtr<WebrtcAudioConduit> self = this;
+  NS_DispatchToMainThread(media::NewRunnableFrom([self]() mutable {
+    MOZ_ASSERT(NS_IsMainThread());
+    if (self->mRtcpEventObserver) {
+      self->mRtcpEventObserver->OnRtcpBye();
+    }
+    return NS_OK;
+  }));
+}
+
+void WebrtcAudioConduit::OnRtcpTimeout() {
+  RefPtr<WebrtcAudioConduit> self = this;
+  NS_DispatchToMainThread(media::NewRunnableFrom([self]() mutable {
+    MOZ_ASSERT(NS_IsMainThread());
+    if (self->mRtcpEventObserver) {
+      self->mRtcpEventObserver->OnRtcpTimeout();
+    }
+    return NS_OK;
+  }));
+}
+
+void WebrtcAudioConduit::SetRtcpEventObserver(
+    mozilla::RtcpEventObserver* observer) {
+  MOZ_ASSERT(NS_IsMainThread());
+  mRtcpEventObserver = observer;
+}
+
 void WebrtcAudioConduit::GetRtpSources(
     const int64_t aTimeNow, nsTArray<dom::RTCRtpSourceEntry>& outSources) {
   MOZ_ASSERT(NS_IsMainThread());
@@ -1080,6 +1108,7 @@ MediaConduitErrorCode WebrtcAudioConduit::CreateChannels() {
   }
 
   mRecvChannelProxy->SetRtpPacketObserver(this);
+  mRecvChannelProxy->SetRtcpEventObserver(this);
   mRecvChannelProxy->RegisterTransport(this);
 
   mSendChannelProxy = vei->GetChannelProxy(mSendChannel);

media/webrtc/signaling/src/media-conduit/AudioConduit.h

@@ -35,6 +35,7 @@ DOMHighResTimeStamp NTPtoDOMHighResTimeStamp(uint32_t ntpHigh, uint32_t ntpLow);
  */
 class WebrtcAudioConduit : public AudioSessionConduit,
                            public webrtc::Transport,
+                           public webrtc::RtcpEventObserver,
                            public webrtc::RtpPacketObserver {
  public:
   // VoiceEngine defined constant for Payload Name Size.
@@ -246,6 +247,11 @@ class WebrtcAudioConduit : public AudioSessionConduit,
   void OnRtpPacket(const webrtc::RTPHeader& aRtpHeader,
                    const int64_t aTimestamp, const uint32_t aJitter) override;
 
+  void OnRtcpBye() override;
+  void OnRtcpTimeout() override;
+
+  void SetRtcpEventObserver(mozilla::RtcpEventObserver* observer) override;
+
   // test-only: inserts fake CSRCs and audio level data
   void InsertAudioLevelForContributingSource(const uint32_t aCsrcSource,
                                              const int64_t aTimestamp,
@@ -362,6 +368,9 @@ class WebrtcAudioConduit : public AudioSessionConduit,
 
   // Accessed only on mStsThread
   Maybe<DOMHighResTimeStamp> mLastRtcpReceived;
+
+  // Accessed only on main thread.
+  mozilla::RtcpEventObserver* mRtcpEventObserver = nullptr;
 };
 
 }  // namespace mozilla

media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h

@@ -12,6 +12,7 @@
 #include "mozilla/RefCounted.h"
 #include "mozilla/UniquePtr.h"
 #include "RtpSourceObserver.h"
+#include "RtcpEventObserver.h"
 #include "CodecConfig.h"
 #include "VideoTypes.h"
 #include "MediaConduitErrors.h"
@@ -254,6 +255,8 @@ class MediaSessionConduit {
 
   virtual Maybe<RefPtr<VideoSessionConduit>> AsVideoSessionConduit() = 0;
 
+  virtual void SetRtcpEventObserver(RtcpEventObserver* observer) = 0;
+
   NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaSessionConduit)
 };
 

media/webrtc/signaling/src/media-conduit/RtcpEventObserver.h

@@ -0,0 +1,20 @@
+/* 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/. */
+
+#ifndef RTCP_EVENT_OBSERVER_
+#define RTCP_EVENT_OBSERVER_
+
+namespace mozilla {
+/**
+ * This provides an interface to allow for receiving notifications
+ * of rtcp bye packets and timeouts.
+ */
+class RtcpEventObserver {
+ public:
+  virtual void OnRtcpBye() = 0;
+  virtual void OnRtcpTimeout() = 0;
+};
+
+}  // namespace mozilla
+#endif

media/webrtc/signaling/src/media-conduit/VideoConduit.cpp

@@ -486,6 +486,7 @@ WebrtcVideoConduit::WebrtcVideoConduit(
       mVideoStatsTimer(NS_NewTimer()) {
   mCall->RegisterConduit(this);
   mRecvStreamConfig.renderer = this;
+  mRecvStreamConfig.rtcp_event_observer = this;
 }
 
 WebrtcVideoConduit::~WebrtcVideoConduit() {
@@ -1786,7 +1787,10 @@ void WebrtcVideoConduit::SelectSendResolution(unsigned short width,
     // Limit resolution to max-fs
     if (mCurSendCodecConfig->mEncodingConstraints.maxFs) {
       // max-fs is in macroblocks, convert to pixels
-      max_fs = std::min(max_fs, static_cast<int>(mCurSendCodecConfig->mEncodingConstraints.maxFs * (16 * 16)));
+      max_fs = std::min(
+          max_fs,
+          static_cast<int>(mCurSendCodecConfig->mEncodingConstraints.maxFs *
+                           (16 * 16)));
     }
     mVideoAdapter->OnResolutionFramerateRequest(
         rtc::Optional<int>(), max_fs, std::numeric_limits<int>::max());
@@ -1885,7 +1889,8 @@ MediaConduitErrorCode WebrtcVideoConduit::SendVideoFrame(
                   this, __FUNCTION__, mSendStreamConfig.rtp.ssrcs.front(),
                   mSendStreamConfig.rtp.ssrcs.front());
 
-    if (mUpdateResolution || frame.width() != mLastWidth || frame.height() != mLastHeight) {
+    if (mUpdateResolution || frame.width() != mLastWidth ||
+        frame.height() != mLastHeight) {
       // See if we need to recalculate what we're sending.
       CSFLogVerbose(LOGTAG, "%s: call SelectSendResolution with %ux%u",
                     __FUNCTION__, frame.width(), frame.height());
@@ -2288,6 +2293,34 @@ uint64_t WebrtcVideoConduit::MozVideoLatencyAvg() {
   return mVideoLatencyAvg / sRoundingPadding;
 }
 
+void WebrtcVideoConduit::OnRtcpBye() {
+  RefPtr<WebrtcVideoConduit> self = this;
+  NS_DispatchToMainThread(media::NewRunnableFrom([self]() mutable {
+    MOZ_ASSERT(NS_IsMainThread());
+    if (self->mRtcpEventObserver) {
+      self->mRtcpEventObserver->OnRtcpBye();
+    }
+    return NS_OK;
+  }));
+}
+
+void WebrtcVideoConduit::OnRtcpTimeout() {
+  RefPtr<WebrtcVideoConduit> self = this;
+  NS_DispatchToMainThread(media::NewRunnableFrom([self]() mutable {
+    MOZ_ASSERT(NS_IsMainThread());
+    if (self->mRtcpEventObserver) {
+      self->mRtcpEventObserver->OnRtcpTimeout();
+    }
+    return NS_OK;
+  }));
+}
+
+void WebrtcVideoConduit::SetRtcpEventObserver(
+    mozilla::RtcpEventObserver* observer) {
+  MOZ_ASSERT(NS_IsMainThread());
+  mRtcpEventObserver = observer;
+}
+
 uint64_t WebrtcVideoConduit::CodecPluginID() {
   MOZ_ASSERT(NS_IsMainThread());
 

media/webrtc/signaling/src/media-conduit/VideoConduit.h

@@ -68,6 +68,7 @@ class WebrtcVideoDecoder : public VideoDecoder, public webrtc::VideoDecoder {};
  */
 class WebrtcVideoConduit
     : public VideoSessionConduit,
+      public webrtc::RtcpEventObserver,
       public webrtc::Transport,
       public webrtc::VideoEncoderFactory,
       public rtc::VideoSinkInterface<webrtc::VideoFrame>,
@@ -295,6 +296,12 @@ class WebrtcVideoConduit
     mRecvStreamStats.RecordTelemetry();
   }
 
+  void OnRtcpBye() override;
+
+  void OnRtcpTimeout() override;
+
+  void SetRtcpEventObserver(mozilla::RtcpEventObserver* observer) override;
+
  private:
   // Don't allow copying/assigning.
   WebrtcVideoConduit(const WebrtcVideoConduit&) = delete;
@@ -640,6 +647,9 @@ class WebrtcVideoConduit
 
   // Accessed only on mStsThread
   Maybe<DOMHighResTimeStamp> mLastRtcpReceived;
+
+  // Accessed only on main thread.
+  mozilla::RtcpEventObserver* mRtcpEventObserver = nullptr;
 };
 }  // namespace mozilla
 

media/webrtc/signaling/src/peerconnection/TransceiverImpl.cpp

@@ -59,6 +59,8 @@ TransceiverImpl::TransceiverImpl(
 
   mConduit->SetPCHandle(mPCHandle);
 
+  mConduit->SetRtcpEventObserver(this);
+
   mTransmitPipeline =
       new MediaPipelineTransmit(mPCHandle, mTransportHandler, mMainThread.get(),
                                 mStsThread.get(), IsVideo(), mConduit);
@@ -959,6 +961,7 @@ void TransceiverImpl::Stop() {
 
   if (mConduit) {
     mConduit->DeleteStreams();
+    mConduit->SetRtcpEventObserver(nullptr);
   }
   mConduit = nullptr;
 }
@@ -979,6 +982,10 @@ void TransceiverImpl::GetRtpSources(
   audio_conduit->GetRtpSources(aTimeNow, outSources);
 }
 
+void TransceiverImpl::OnRtcpBye() { SetReceiveTrackMuted(true); }
+
+void TransceiverImpl::OnRtcpTimeout() { SetReceiveTrackMuted(true); }
+
 void TransceiverImpl::InsertAudioLevelForContributingSource(
     const uint32_t aSource, const int64_t aTimestamp,
     const uint32_t aRtpTimestamp, const bool aHasLevel, const uint8_t aLevel) {

media/webrtc/signaling/src/peerconnection/TransceiverImpl.h

@@ -12,6 +12,7 @@
 #include "mozilla/dom/MediaStreamTrack.h"
 #include "ErrorList.h"
 #include "signaling/src/jsep/JsepTransceiver.h"
+#include "signaling/src/media-conduit/RtcpEventObserver.h"
 
 class nsIPrincipal;
 
@@ -43,7 +44,7 @@ struct RTCRtpSourceEntry;
  * Audio/VideoConduit for feeding RTP/RTCP into webrtc.org for decoding, and
  * feeding audio/video frames into webrtc.org for encoding into RTP/RTCP.
  */
-class TransceiverImpl : public nsISupports {
+class TransceiverImpl : public nsISupports, public RtcpEventObserver {
  public:
   /**
    * |aReceiveTrack| is always set; this holds even if the remote end has not
@@ -119,6 +120,10 @@ class TransceiverImpl : public nsISupports {
   void GetRtpSources(const int64_t aTimeNow,
                      nsTArray<dom::RTCRtpSourceEntry>& outSources) const;
 
+  void OnRtcpBye() override;
+
+  void OnRtcpTimeout() override;
+
   // test-only: insert fake CSRCs and audio levels for testing
   void InsertAudioLevelForContributingSource(const uint32_t aSource,
                                              const int64_t aTimestamp,