Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks.

OnTransportOverChanged is merged into OnNetworkRouteChanged in MediaChannel
because the transport overhead will be added to rtc::NetworkRoute structure.

This CL depends on https://webrtc-review.googlesource.com/c/src/+/13520

Bug: None
Change-Id: I6ed6583f6c91db4ce61a89406de39774239f3a04
Reviewed-on: https://webrtc-review.googlesource.com/15200
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20678}

media/base/fakemediaengine.h

@@ -273,14 +273,12 @@ template <class Base> class RtpHelper : public Base {
   virtual void OnReadyToSend(bool ready) {
     ready_to_send_ = ready;
   }
-  virtual void OnTransportOverheadChanged(int transport_overhead_per_packet) {
-    transport_overhead_per_packet_ = transport_overhead_per_packet;
-  }
 
   virtual void OnNetworkRouteChanged(const std::string& transport_name,
                                      const rtc::NetworkRoute& network_route) {
     last_network_route_ = network_route;
     ++num_network_route_changes_;
+    transport_overhead_per_packet_ = network_route.packet_overhead;
   }
   bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
   bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }

media/base/mediachannel.h

@@ -415,9 +415,6 @@ class MediaChannel : public sigslot::has_slots<> {
   virtual void OnNetworkRouteChanged(
       const std::string& transport_name,
       const rtc::NetworkRoute& network_route) = 0;
-  // Called when the rtp transport overhead changed.
-  virtual void OnTransportOverheadChanged(
-      int transport_overhead_per_packet) = 0;
   // Creates a new outgoing media stream with SSRCs and CNAME as described
   // by sp.
   virtual bool AddSendStream(const StreamParams& sp) = 0;

media/base/rtpdataengine.h

@@ -84,7 +84,6 @@ class RtpDataMediaChannel : public DataMediaChannel {
   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                               const rtc::PacketTime& packet_time) {}
   virtual void OnReadyToSend(bool ready) {}
-  virtual void OnTransportOverheadChanged(int transport_overhead_per_packet) {}
   virtual bool SendData(
     const SendDataParams& params,
     const rtc::CopyOnWriteBuffer& payload,

media/engine/webrtcvideoengine.cc

@@ -1470,13 +1470,10 @@ void WebRtcVideoChannel::OnReadyToSend(bool ready) {
 void WebRtcVideoChannel::OnNetworkRouteChanged(
     const std::string& transport_name,
     const rtc::NetworkRoute& network_route) {
+  // TODO(zhihaung): Merge these two callbacks.
   call_->OnNetworkRouteChanged(transport_name, network_route);
-}
-
-void WebRtcVideoChannel::OnTransportOverheadChanged(
-    int transport_overhead_per_packet) {
   call_->OnTransportOverheadChanged(webrtc::MediaType::VIDEO,
-                                    transport_overhead_per_packet);
+                                    network_route.packet_overhead);
 }
 
 void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) {

media/engine/webrtcvideoengine.h

@@ -167,7 +167,6 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
   void OnReadyToSend(bool ready) override;
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override;
-  void OnTransportOverheadChanged(int transport_overhead_per_packet) override;
   void SetInterface(NetworkInterface* iface) override;
 
   // Implemented for VideoMediaChannelTest.

media/engine/webrtcvoiceengine.cc

@@ -2159,7 +2159,11 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
 void WebRtcVoiceMediaChannel::OnNetworkRouteChanged(
     const std::string& transport_name,
     const rtc::NetworkRoute& network_route) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  // TODO(zhihaung): Merge these two callbacks.
   call_->OnNetworkRouteChanged(transport_name, network_route);
+  call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO,
+                                    network_route.packet_overhead);
 }
 
 bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
@@ -2205,13 +2209,6 @@ void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready) {
       ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
-void WebRtcVoiceMediaChannel::OnTransportOverheadChanged(
-    int transport_overhead_per_packet) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO,
-                                    transport_overhead_per_packet);
-}
-
 bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());

media/engine/webrtcvoiceengine.h

@@ -207,7 +207,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override;
   void OnReadyToSend(bool ready) override;
-  void OnTransportOverheadChanged(int transport_overhead_per_packet) override;
   bool GetStats(VoiceMediaInfo* info) override;
 
   // SSRC=0 will set the audio sink on the latest unsignaled stream, future or

pc/channel.cc

@@ -595,13 +595,8 @@ void BaseChannel::OnNetworkRouteChanged(
   RTC_DCHECK(network_thread_->IsCurrent());
   rtc::NetworkRoute new_route;
   if (network_route) {
-    invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [=] {
-      media_channel_->OnTransportOverheadChanged(
-          network_route->packet_overhead);
-    });
     new_route = *(network_route);
   }
-
   // Note: When the RTCP-muxing is not enabled, RTCP transport and RTP transport
   // use the same transport name and MediaChannel::OnNetworkRouteChanged cannot
   // work correctly. Intentionally leave it broken to simplify the code and