Remove audio/video distinction for probe packets.

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 261 matching lines @@
   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), false))
+                             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));
 }
 
@@ skipping 56 matching lines @@
 TEST(AudioReceiveStreamTest, SetGain) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state());
   EXPECT_CALL(*helper.channel_proxy(),
       SetChannelOutputVolumeScaling(FloatEq(0.765f)));
   recv_stream.SetGain(0.765f);
 }
 }  // namespace test
 }  // namespace webrtc