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

webrtc/media/engine/fakewebrtccall.h

Index: webrtc/media/engine/fakewebrtccall.h
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
index 15f2108cca230777460486fecf2a632993b7fce2..7f3a3ae0c3a93c0bbd90cb1806f36d603e1d8a28 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -47,7 +47,9 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
       int id, const webrtc::AudioSendStream::Config& config);
 
   int id() const { return id_; }
-  const webrtc::AudioSendStream::Config& GetConfig() const;
+  const webrtc::AudioSendStream::Config& GetConfig() const override;
+  webrtc::RtpState GetRtpState() const override;
+  const webrtc::TimeInterval& GetActiveLifetime() const override;
   void SetStats(const webrtc::AudioSendStream::Stats& stats);
   TelephoneEvent GetLatestTelephoneEvent() const;
   bool IsSending() const { return sending_; }