| Index: webrtc/media/engine/fakewebrtccall.cc
|
| diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
|
| index 2e793b85253800211b4ca7414ccdeed43aa52df3..e865fe12b18be7ba2bb3dbe9e441e719715b28be 100644
|
| --- a/webrtc/media/engine/fakewebrtccall.cc
|
| +++ b/webrtc/media/engine/fakewebrtccall.cc
|
| @@ -36,6 +36,15 @@ const webrtc::AudioSendStream::Config&
|
| 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;
|
|
|