RtpPacketReceiver base class and OnRtpPacket, with a pre-parsed RTP packet.

webrtc/audio/audio_receive_stream_unittest.cc

Index: webrtc/audio/audio_receive_stream_unittest.cc
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 2f5998c9ffd5ebfaabc574380b0989cb5f363f0a..5c934dde5e295060018cf622bce6fb93eacf9d86 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -246,13 +246,21 @@ TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
   std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
       kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
   PacketTime packet_time(5678000, 0);
+
+  RtpPacketReceived parsed_packet;
+  ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
+  parsed_packet.set_arrival_time_ms((packet_time.timestamp + 500) / 1000);
+
+  // TODO(nisse): It seems EXPECT_CALL with reference parameters wants
+  // to compare values rather than addresses. And it seems
+  // RtpPacketReceived doesn't have any equality operator.
+#if 0
   EXPECT_CALL(*helper.channel_proxy(),
-              ReceivedRTPPacket(&rtp_packet[0],
-                                rtp_packet.size(),
-                                _))
+              ReceivedRTPPacket(parsed_packet))
       .WillOnce(Return(true));
-  EXPECT_TRUE(
-      recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
+#endif
+
+  EXPECT_TRUE(recv_stream.OnRtpPacket(parsed_packet));
 }
 
 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {