Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

webrtc/media/engine/fakewebrtccall.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 18 matching lines @@
 void FakeAudioSendStream::Reconfigure(
     const webrtc::AudioSendStream::Config& config) {
   config_ = config;
 }
 
 const webrtc::AudioSendStream::Config&
     FakeAudioSendStream::GetConfig() const {
   return config_;
 }
 
+webrtc::RtpState FakeAudioSendStream::GetRtpState() const {
+  return webrtc::RtpState();
+}
+
+const webrtc::TimeInterval& FakeAudioSendStream::GetActiveLifetime() const {
+  static webrtc::TimeInterval fake_time_interval;
+  return fake_time_interval;
+}
+
 void FakeAudioSendStream::SetStats(
     const webrtc::AudioSendStream::Stats& stats) {
   stats_ = stats;
 }
 
 FakeAudioSendStream::TelephoneEvent
     FakeAudioSendStream::GetLatestTelephoneEvent() const {
   return latest_telephone_event_;
 }
 
@@ skipping 582 matching lines @@
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
   if (sent_packet.packet_id >= 0) {
     last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
   }
 }
 
 }  // namespace cricket