Always call RemoteBitrateEstimator::IncomingPacket from Call.

webrtc/audio/audio_receive_stream_unittest.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 230 matching lines @@
 }
 
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.packet_router(),
       helper.remote_bitrate_estimator(),
       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;
   helper.SetupMockForBweFeedback(true);
   internal::AudioReceiveStream recv_stream(
       helper.packet_router(),
       helper.remote_bitrate_estimator(),
       helper.config(), helper.audio_state(), helper.event_log());
   const int kTransportSequenceNumberValue = 1234;
   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(),
                                 _))
       .WillOnce(Return(true));
   EXPECT_TRUE(
       recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
 }
 
 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
@@ skipping 91 matching lines @@
 
   EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0));
   EXPECT_CALL(helper.voice_engine(), StopPlayout(_));
   EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream))
       .WillOnce(Return(true));
 
   recv_stream.Start();
 }
 }  // namespace test
 }  // namespace webrtc