Always call RemoteBitrateEstimator::IncomingPacket from Call.

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 4c443e0f2e99c11391713c0f789326aef2ae2c9f..8df66fe127a55340b6cc079a98a17dd2e504ad76 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -248,13 +248,6 @@ TEST(AudioReceiveStreamTest, ConstructDestruct) {
       helper.config(), helper.audio_state(), helper.event_log());
 }
 
-MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
-  return arg.extension.hasTransportSequenceNumber ==
-             expected_extension.hasTransportSequenceNumber &&
-         arg.extension.transportSequenceNumber ==
-             expected_extension.transportSequenceNumber;
-}
-
 TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
   ConfigHelper helper;
   helper.config().rtp.transport_cc = true;
@@ -267,15 +260,6 @@ TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
   std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
       kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
   PacketTime packet_time(5678000, 0);
-  const size_t kExpectedHeaderLength = 20;
-  RTPHeaderExtension expected_extension;
-  expected_extension.hasTransportSequenceNumber = true;
-  expected_extension.transportSequenceNumber = kTransportSequenceNumberValue;
-  EXPECT_CALL(*helper.remote_bitrate_estimator(),
-              IncomingPacket(packet_time.timestamp / 1000,
-                             rtp_packet.size() - kExpectedHeaderLength,
-                             VerifyHeaderExtension(expected_extension)))
-      .Times(1);
   EXPECT_CALL(*helper.channel_proxy(),
               ReceivedRTPPacket(&rtp_packet[0],
                                 rtp_packet.size(),