Made the AudioProcessing class a pure interface.

webrtc/media/engine/fakewebrtcvoiceengine.h

Index: webrtc/media/engine/fakewebrtcvoiceengine.h
diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 0526b3995e238dcb3d648e361c828f2c05e5049b..e0e71fb8002cd8eff7e37b6257dd575ff1157fb5 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -116,6 +116,8 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
   WEBRTC_STUB(StartDebugRecording,
               (const char filename[kMaxFilenameSize], int64_t max_size_bytes));
   WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes));
+  WEBRTC_STUB(StartDebugRecording, (FILE * handle));
+  WEBRTC_STUB(StartDebugRecordingForPlatformFile, (rtc::PlatformFile handle));
   WEBRTC_STUB(StopDebugRecording, ());
   WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
   webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }