Bug 1404997 - P11. More constness fixes. r=pehrsons

MozReview-Commit-ID: 5qgO43V9R9P

--HG--
extra : rebase_source : 6f387e73486079737c066420a3e95f15c1c0ac16

media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp

@@ -335,18 +335,16 @@ protected:
     ImageFormat format = aImage->GetFormat();
     if (format == ImageFormat::PLANAR_YCBCR) {
       // Cast away constness b/c some of the accessors are non-const
-      PlanarYCbCrImage* yuv = const_cast<PlanarYCbCrImage*>(
+      const PlanarYCbCrData* data =
-        static_cast<const PlanarYCbCrImage*>(aImage));
+        static_cast<const PlanarYCbCrImage*>(aImage)->GetData();
-
-      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 width = aImage->GetSize().width;
-        uint32_t height = yuv->GetSize().height;
+        uint32_t height = aImage->GetSize().height;
 
         rtc::Callback0<void> callback_unused;
         rtc::scoped_refptr<webrtc::WrappedI420Buffer> video_frame_buffer(
@@ -594,7 +592,7 @@ protected:
   }
 
   RefPtr<AudioSessionConduit> mConduit;
-  RefPtr<AutoTaskQueue> mTaskQueue;
+  const RefPtr<AutoTaskQueue> mTaskQueue;
   // Only accessed on mTaskQueue
   UniquePtr<AudioPacketizer<int16_t, int16_t>> mPacketizer;
   // A buffer to hold a single packet of audio.
@@ -625,7 +623,6 @@ MediaPipeline::MediaPipeline(const std::string& aPc,
   , mRtpBytesSent(0)
   , mRtpBytesReceived(0)
   , mPc(aPc)
-  , mDescription()
   , mRtpParser(webrtc::RtpHeaderParser::Create())
   , mPacketDumper(new PacketDumper(mPc))
 {
@@ -989,7 +986,7 @@ MediaPipeline::UpdateRtcpMuxState(TransportInfo& aInfo)
 }
 
 nsresult
-MediaPipeline::SendPacket(TransportFlow* aFlow, const void* aData, int aLen)
+MediaPipeline::SendPacket(const TransportFlow* aFlow, const void* aData, int aLen)
 {
   ASSERT_ON_THREAD(mStsThread);
 
@@ -1273,7 +1270,7 @@ MediaPipeline::RtcpPacketReceived(TransportLayer* aLayer,
 }
 
 bool
-MediaPipeline::IsRtp(const unsigned char* aData, size_t aLen)
+MediaPipeline::IsRtp(const unsigned char* aData, size_t aLen) const
 {
   if (aLen < 2)
     return false;
@@ -1523,8 +1520,14 @@ MediaPipelineTransmit::MediaPipelineTransmit(
                   aMainThread,
                   aStsThread,
                   aConduit)
-  , mListener(new PipelineListener(aConduit))
   , mIsVideo(aIsVideo)
+  , mListener(new PipelineListener(aConduit))
+  , mFeeder(aIsVideo ? MakeAndAddRef<VideoFrameFeeder>(mListener)
+                     : nullptr) // For video we send frames to an
+                                // async VideoFrameConverter that
+                                // calls back to a VideoFrameFeeder
+                                // that feeds I420 frames to
+                                // VideoConduit.
   , mDomTrack(aDomTrack)
   , mTransmitting(false)
 {
@@ -1534,14 +1537,8 @@ MediaPipelineTransmit::MediaPipelineTransmit(
       static_cast<AudioSessionConduit*>(aConduit.get()));
     mListener->SetAudioProxy(mAudioProcessing);
   } else { // Video
-    // For video we send frames to an async VideoFrameConverter that calls
-    // back to a VideoFrameFeeder that feeds I420 frames to VideoConduit.
-
-    mFeeder = MakeAndAddRef<VideoFrameFeeder>(mListener);
-
     mConverter = MakeAndAddRef<VideoFrameConverter>();
     mConverter->AddListener(mFeeder);
-
     mListener->SetVideoFrameConverter(mConverter);
   }
 }
@@ -1650,7 +1647,7 @@ MediaPipelineTransmit::IsVideo() const
 }
 
 void
-MediaPipelineTransmit::UpdateSinkIdentity_m(MediaStreamTrack* aTrack,
+MediaPipelineTransmit::UpdateSinkIdentity_m(const MediaStreamTrack* aTrack,
                                             nsIPrincipal* aPrincipal,
                                             const PeerIdentity* aSinkIdentity)
 {
@@ -2022,17 +2019,14 @@ MediaPipelineTransmit::PipelineListener::NewData(const MediaSegment& aMedia,
   if (aMedia.GetType() == MediaSegment::AUDIO) {
     MOZ_RELEASE_ASSERT(aRate > 0);
 
-    AudioSegment* audio =
+    const AudioSegment* audio = static_cast<const AudioSegment*>(&aMedia);
-      const_cast<AudioSegment*>(static_cast<const AudioSegment*>(&aMedia));
+    for (AudioSegment::ConstChunkIterator iter(*audio); !iter.IsEnded();
-    for (AudioSegment::ChunkIterator iter(*audio); !iter.IsEnded();
          iter.Next()) {
       mAudioProcessing->QueueAudioChunk(aRate, *iter, mEnabled);
     }
   } else {
-    VideoSegment* video =
+    const VideoSegment* video = static_cast<const VideoSegment*>(&aMedia);
-      const_cast<VideoSegment*>(static_cast<const VideoSegment*>(&aMedia));
+    for (VideoSegment::ConstChunkIterator iter(*video); !iter.IsEnded();
-    VideoSegment::ChunkIterator iter(*video);
-    for (VideoSegment::ChunkIterator iter(*video); !iter.IsEnded();
          iter.Next()) {
       mConverter->QueueVideoChunk(*iter, !mEnabled);
     }
@@ -2070,7 +2064,7 @@ public:
   void TrackAdded(TrackTicks aTime);
 
 private:
-  RefPtr<GenericReceiveListener> mListener;
+  const RefPtr<GenericReceiveListener> mListener;
 };
 
 class GenericReceiveListener : public MediaStreamListener
@@ -2170,7 +2164,7 @@ public:
         mListener->SetPrincipalHandle_msg(mPrincipalHandle);
       }
 
-      RefPtr<GenericReceiveListener> mListener;
+      const RefPtr<GenericReceiveListener> mListener;
       PrincipalHandle mPrincipalHandle;
     };
 

media/webrtc/signaling/src/mediapipeline/MediaPipeline.h

@@ -155,7 +155,7 @@ public:
 
   private:
     static const double constexpr EXPIRY_TIME_MILLISECONDS = 10 * 1000;
-    uint32_t mCsrc;
+    const uint32_t mCsrc;
     DOMHighResTimeStamp mTimestamp;
   };
 
@@ -203,7 +203,7 @@ public:
 
     // Creates a cycle, which we break with Detach
     RefPtr<MediaPipeline> mPipeline;
-    nsCOMPtr<nsIEventTarget> mStsThread;
+    const nsCOMPtr<nsIEventTarget> mStsThread;
   };
 
 protected:
@@ -250,10 +250,12 @@ protected:
   virtual void OnRtpPacketReceived() {};
   void IncrementRtcpPacketsReceived();
 
-  virtual nsresult SendPacket(TransportFlow* aFlow, const void* aData, int aLen);
+  virtual nsresult SendPacket(const TransportFlow* aFlow,
+                              const void* aData,
+                              int aLen);
 
   // Process slots on transports
-  void StateChange(TransportFlow* flow, TransportLayer::State);
+  void StateChange(TransportFlow* aFlow, TransportLayer::State);
   void RtpPacketReceived(TransportLayer* aLayer,
                          const unsigned char* aData,
                          size_t aLen);
@@ -264,7 +266,7 @@ protected:
                       const unsigned char* aData,
                       size_t aLen);
 
-  DirectionType mDirection;
+  const DirectionType mDirection;
   size_t mLevel;
   RefPtr<MediaSessionConduit> mConduit; // Our conduit. Written on the main
                                         // thread. Read on STS thread.
@@ -275,8 +277,8 @@ protected:
 
   // Pointers to the threads we need. Initialized at creation
   // and used all over the place.
-  nsCOMPtr<nsIEventTarget> mMainThread;
+  const nsCOMPtr<nsIEventTarget> mMainThread;
-  nsCOMPtr<nsIEventTarget> mStsThread;
+  const nsCOMPtr<nsIEventTarget> mStsThread;
 
   // Created in c'tor. Referenced by the conduit.
   RefPtr<PipelineTransport> mTransport;
@@ -294,12 +296,12 @@ protected:
   std::map<uint32_t, RtpCSRCStats> mCsrcStats;
 
   // Written in c'tor. Read on STS thread.
-  std::string mPc;
+  const std::string mPc;
   std::string mDescription;
 
   // Written in c'tor, all following accesses are on the STS thread.
   nsAutoPtr<MediaPipelineFilter> mFilter;
-  nsAutoPtr<webrtc::RtpHeaderParser> mRtpParser;
+  const nsAutoPtr<webrtc::RtpHeaderParser> mRtpParser;
 
   nsAutoPtr<PacketDumper> mPacketDumper;
 
@@ -307,7 +309,7 @@ private:
   // Gets the current time as a DOMHighResTimeStamp
   static DOMHighResTimeStamp GetNow();
 
-  bool IsRtp(const unsigned char* aData, size_t aLen);
+  bool IsRtp(const unsigned char* aData, size_t aLen) const;
   // Must be called on the STS thread.  Must be called after DetachMedia().
   void DetachTransport_s();
 };
@@ -325,7 +327,7 @@ public:
   NS_IMETHOD Run() override { return NS_OK; }
 
 private:
-  RefPtr<MediaSessionConduit> mConduit;
+  const RefPtr<MediaSessionConduit> mConduit;
 };
 
 // A specialization of pipeline for reading from an input device
@@ -350,7 +352,7 @@ public:
   // 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 `mDomTrack` for us to apply the update.
-  virtual void UpdateSinkIdentity_m(dom::MediaStreamTrack* aTrack,
+  virtual void UpdateSinkIdentity_m(const dom::MediaStreamTrack* aTrack,
                                     nsIPrincipal* aPrincipal,
                                     const PeerIdentity* aSinkIdentity);
 
@@ -376,11 +378,11 @@ protected:
   void SetDescription();
 
 private:
-  RefPtr<PipelineListener> mListener;
+  const bool mIsVideo;
+  const RefPtr<PipelineListener> mListener;
+  const RefPtr<VideoFrameFeeder> mFeeder;
   RefPtr<AudioProxyThread> mAudioProcessing;
-  RefPtr<VideoFrameFeeder> mFeeder;
   RefPtr<VideoFrameConverter> mConverter;
-  bool mIsVideo;
   RefPtr<dom::MediaStreamTrack> mDomTrack;
   bool mTransmitting;
 };
@@ -464,7 +466,7 @@ private:
   // Separate class to allow ref counting
   class PipelineListener;
 
-  RefPtr<PipelineRenderer> mRenderer;
+  const RefPtr<PipelineRenderer> mRenderer;
   RefPtr<PipelineListener> mListener;
 };
 

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

@@ -1503,7 +1503,7 @@ PeerConnectionMedia::UpdateRemoteStreamPrincipals_m(nsIPrincipal* aPrincipal)
 }
 
 void
-PeerConnectionMedia::UpdateSinkIdentity_m(MediaStreamTrack* aTrack,
+PeerConnectionMedia::UpdateSinkIdentity_m(const MediaStreamTrack* aTrack,
                                           nsIPrincipal* aPrincipal,
                                           const PeerIdentity* aSinkIdentity)
 {

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

@@ -140,7 +140,7 @@ class PeerConnectionMedia : public sigslot::has_slots<> {
   // black/silence.  Once the peer is identified, re-enable those streams.
   // aTrack will be set if this update came from a principal change on aTrack.
   // TODO: Move to PeerConnectionImpl
-  void UpdateSinkIdentity_m(dom::MediaStreamTrack* aTrack,
+  void UpdateSinkIdentity_m(const dom::MediaStreamTrack* aTrack,
                             nsIPrincipal* aPrincipal,
                             const PeerIdentity* aSinkIdentity);
   // this determines if any track is peerIdentity constrained

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

@@ -120,7 +120,7 @@ TransceiverImpl::InitVideo()
 }
 
 nsresult
-TransceiverImpl::UpdateSinkIdentity(dom::MediaStreamTrack* aTrack,
+TransceiverImpl::UpdateSinkIdentity(const dom::MediaStreamTrack* aTrack,
                                     nsIPrincipal* aPrincipal,
                                     const PeerIdentity* aSinkIdentity)
 {

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

@@ -65,7 +65,7 @@ public:
 
   nsresult UpdateSendTrack(dom::MediaStreamTrack* aSendTrack);
 
-  nsresult UpdateSinkIdentity(dom::MediaStreamTrack* aTrack,
+  nsresult UpdateSinkIdentity(const dom::MediaStreamTrack* aTrack,
                               nsIPrincipal* aPrincipal,
                               const PeerIdentity* aSinkIdentity);