Implement AudioReceiveStream::GetStats().

talk/media/webrtc/fakewebrtccall.cc

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 22 matching lines @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
 
 namespace cricket {
 FakeAudioReceiveStream::FakeAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config)
     : config_(config), received_packets_(0) {
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
-webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
-  return webrtc::AudioReceiveStream::Stats();
-}
-
 const webrtc::AudioReceiveStream::Config&
     FakeAudioReceiveStream::GetConfig() const {
   return config_;
 }
 
+void FakeAudioReceiveStream::SetStats(
+    const webrtc::AudioReceiveStream::Stats& stats) {
+  stats_ = stats;
+}
+
 void FakeAudioReceiveStream::IncrementReceivedPackets() {
   received_packets_++;
 }
 
 FakeVideoSendStream::FakeVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const webrtc::VideoEncoderConfig& encoder_config)
     : sending_(false),
       config_(config),
       codec_settings_set_(false),
@@ skipping 302 matching lines @@
 }
 
 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
   network_state_ = state;
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
 }
 }  // namespace cricket