Bug 1225722 - Implement getters in the conduits for currently active payload types. r=bwc

Differential Revision: https://phabricator.services.mozilla.com/D135866

dom/media/webrtc/libwebrtcglue/AudioConduit.cpp

@@ -875,4 +875,34 @@ void WebrtcAudioConduit::DeliverPacket(rtc::CopyOnWriteBuffer packet,
   }
 }
 
+Maybe<int> WebrtcAudioConduit::ActiveSendPayloadType() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  auto stats = GetSenderStats();
+  if (!stats) {
+    return Nothing();
+  }
+
+  if (!stats->codec_payload_type) {
+    return Nothing();
+  }
+
+  return Some(*stats->codec_payload_type);
+}
+
+Maybe<int> WebrtcAudioConduit::ActiveRecvPayloadType() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  auto stats = GetReceiverStats();
+  if (!stats) {
+    return Nothing();
+  }
+
+  if (!stats->codec_payload_type) {
+    return Nothing();
+  }
+
+  return Some(*stats->codec_payload_type);
+}
+
 }  // namespace mozilla

dom/media/webrtc/libwebrtcglue/AudioConduit.h

@@ -29,6 +29,9 @@ struct DtmfEvent;
 class WebrtcAudioConduit : public AudioSessionConduit,
                            public webrtc::RtcpEventObserver {
  public:
+  Maybe<int> ActiveSendPayloadType() const override;
+  Maybe<int> ActiveRecvPayloadType() const override;
+
   void OnRtpReceived(MediaPacket&& aPacket, webrtc::RTPHeader&& aHeader);
   void OnRtcpReceived(MediaPacket&& aPacket);
 

dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h

@@ -115,6 +115,10 @@ class MediaSessionConduit {
 
   virtual Type type() const = 0;
 
+  // Call thread only
+  virtual Maybe<int> ActiveSendPayloadType() const = 0;
+  virtual Maybe<int> ActiveRecvPayloadType() const = 0;
+
   // Whether transport is currently sending and receiving packets
   virtual void SetTransportActive(bool aActive) = 0;
 

dom/media/webrtc/libwebrtcglue/VideoConduit.cpp

@@ -1892,4 +1892,33 @@ bool WebrtcVideoConduit::HasH264Hardware() {
          status == nsIGfxInfo::FEATURE_STATUS_OK;
 }
 
+Maybe<int> WebrtcVideoConduit::ActiveSendPayloadType() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  if (!mSendStream) {
+    return Nothing();
+  }
+
+  if (mSendStreamConfig.rtp.payload_type == -1) {
+    return Nothing();
+  }
+
+  return Some(mSendStreamConfig.rtp.payload_type);
+}
+
+Maybe<int> WebrtcVideoConduit::ActiveRecvPayloadType() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  auto stats = GetReceiverStats();
+  if (!stats) {
+    return Nothing();
+  }
+
+  if (stats->current_payload_type == -1) {
+    return Nothing();
+  }
+
+  return Some(stats->current_payload_type);
+}
+
 }  // namespace mozilla

dom/media/webrtc/libwebrtcglue/VideoConduit.h

@@ -71,6 +71,9 @@ class WebrtcVideoConduit
   // Returns true when both encoder and decoder are HW accelerated.
   static bool HasH264Hardware();
 
+  Maybe<int> ActiveSendPayloadType() const override;
+  Maybe<int> ActiveRecvPayloadType() const override;
+
   /**
    * Function to attach Renderer end-point for the Media-Video conduit.
    * @param aRenderer : Reference to the concrete mozilla Video renderer

media/webrtc/signaling/gtest/MockConduit.h

@@ -24,6 +24,8 @@ class MockConduit : public MediaSessionConduit {
   MockConduit() = default;
 
   MOCK_CONST_METHOD0(type, Type());
+  MOCK_CONST_METHOD0(ActiveSendPayloadType, Maybe<int>());
+  MOCK_CONST_METHOD0(ActiveRecvPayloadType, Maybe<int>());
   MOCK_METHOD1(SetTransportActive, void(bool));
   MOCK_METHOD0(SenderRtpSendEvent, MediaEventSourceExc<MediaPacket>&());
   MOCK_METHOD0(SenderRtcpSendEvent, MediaEventSourceExc<MediaPacket>&());