Moves conversion to ReceivedPacket from RtpPacketReceived to Call.

This moves the conversion from RtpPacketReceived to ReceivedPacket to
Call rather than RtpTransportController. This prepares for reusing the
struct for receive side network state estimation.

Bug: webrtc:10742
Change-Id: I9581438bc912ef4bb635a5d9a6dea488cf871d48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141872
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28284}
diff --git a/call/call.cc b/call/call.cc
index e50ac97..a25e161 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -1497,7 +1497,10 @@
   RTPHeader header;
   packet.GetHeader(&header);
 
-  transport_send_ptr_->OnReceivedPacket(packet);
+  ReceivedPacket packet_msg;
+  packet_msg.size = DataSize::bytes(packet.payload_size());
+  packet_msg.receive_time = Timestamp::ms(packet.arrival_time_ms());
+  transport_send_ptr_->OnReceivedPacket(packet_msg);
 
   if (!use_send_side_bwe && header.extension.hasTransportSequenceNumber) {
     // Inconsistent configuration of send side BWE. Do nothing.
diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
index a4a5dda..7ff6525 100644
--- a/call/rtp_transport_controller_send.cc
+++ b/call/rtp_transport_controller_send.cc
@@ -327,10 +327,7 @@
 }
 
 void RtpTransportControllerSend::OnReceivedPacket(
-    const RtpPacketReceived& received_packet) {
-  ReceivedPacket packet_msg;
-  packet_msg.size = DataSize::bytes(received_packet.payload_size());
-  packet_msg.receive_time = Timestamp::ms(received_packet.arrival_time_ms());
+    const ReceivedPacket& packet_msg) {
   task_queue_.PostTask([this, packet_msg]() {
     RTC_DCHECK_RUN_ON(&task_queue_);
     if (controller_)
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index 2ba566a..10af91d 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -96,7 +96,7 @@
   int64_t GetFirstPacketTimeMs() const override;
   void EnablePeriodicAlrProbing(bool enable) override;
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
-  void OnReceivedPacket(const RtpPacketReceived& received_packet) override;
+  void OnReceivedPacket(const ReceivedPacket& packet_msg) override;
 
   void SetSdpBitrateParameters(const BitrateConstraints& constraints) override;
   void SetClientBitratePreferences(const BitrateSettings& preferences) override;
diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index 597e934..860b705 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -149,7 +149,7 @@
   virtual int64_t GetFirstPacketTimeMs() const = 0;
   virtual void EnablePeriodicAlrProbing(bool enable) = 0;
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
-  virtual void OnReceivedPacket(const RtpPacketReceived& received_packet) = 0;
+  virtual void OnReceivedPacket(const ReceivedPacket& received_packet) = 0;
 
   virtual void SetSdpBitrateParameters(
       const BitrateConstraints& constraints) = 0;
diff --git a/call/test/mock_rtp_transport_controller_send.h b/call/test/mock_rtp_transport_controller_send.h
index 38fbb93..98f4251 100644
--- a/call/test/mock_rtp_transport_controller_send.h
+++ b/call/test/mock_rtp_transport_controller_send.h
@@ -65,7 +65,7 @@
   MOCK_METHOD1(SetSdpBitrateParameters, void(const BitrateConstraints&));
   MOCK_METHOD1(SetClientBitratePreferences, void(const BitrateSettings&));
   MOCK_METHOD1(OnTransportOverheadChanged, void(size_t));
-  MOCK_METHOD1(OnReceivedPacket, void(const RtpPacketReceived&));
+  MOCK_METHOD1(OnReceivedPacket, void(const ReceivedPacket&));
 };
 }  // namespace webrtc
 #endif  // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_