Initial implementation of RtpTransportControllerReceive and related interfaces.

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 f600b857b634ed0a19ef48d15bf745c76484cb61..700889a80eab81219d726a00397270c2911097b0 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -89,12 +89,6 @@ struct ConfigHelper {
           EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
           EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
           EXPECT_CALL(*channel_proxy_,
-              SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
-                  .Times(1);
-          EXPECT_CALL(*channel_proxy_,
-              EnableReceiveTransportSequenceNumber(kTransportSequenceNumberId))
-                  .Times(1);
-          EXPECT_CALL(*channel_proxy_,
               RegisterReceiverCongestionControlObjects(&packet_router_))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_, ResetReceiverCongestionControlObjects())
@@ -260,7 +254,7 @@ TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
   EXPECT_CALL(*helper.channel_proxy(),
               OnRtpPacket(testing::Ref(parsed_packet)));
 
-  recv_stream.OnRtpPacket(parsed_packet);
+  EXPECT_TRUE(recv_stream.OnRtpPacketReceive(&parsed_packet));
 }
 
 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {