| Index: webrtc/media/engine/fakewebrtccall.cc
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| diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
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| index 2e793b85253800211b4ca7414ccdeed43aa52df3..e865fe12b18be7ba2bb3dbe9e441e719715b28be 100644
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| --- a/webrtc/media/engine/fakewebrtccall.cc
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| +++ b/webrtc/media/engine/fakewebrtccall.cc
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| @@ -36,6 +36,15 @@ const webrtc::AudioSendStream::Config&
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|    return config_;
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|  }
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|  
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| +webrtc::RtpState FakeAudioSendStream::GetRtpState() const {
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| +  return webrtc::RtpState();
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| +}
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| +
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| +const webrtc::TimeInterval& FakeAudioSendStream::GetActiveLifetime() const {
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| +  static webrtc::TimeInterval fake_time_interval;
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| +  return fake_time_interval;
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| +}
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| +
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|  void FakeAudioSendStream::SetStats(
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|      const webrtc::AudioSendStream::Stats& stats) {
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|    stats_ = stats;
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| 
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