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 0239dd389c28c5a953573c86d500fdb386af924b..11e3a8b4056015e97c1c88edb06fba44935aa49d 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);
+  recv_stream.OnRtpPacketReceive(&parsed_packet);

[the sun, 2017/04/18 09:59:00]

EXPECT_TRUE(

[nisse-webrtc, 2017/04/19 08:35:43]

Done.

 }
 
 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {