Added option to specify a maximum file size when recording an AEC dump.

talk/media/webrtc/fakewebrtcvoiceengine.h

Index: talk/media/webrtc/fakewebrtcvoiceengine.h
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 1167b6b9830c2f9edd1ff0b9caa606a50814d372..0244b287526704d18f6f56e2448468bd726d5ef2 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -157,7 +157,10 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
   WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
   WEBRTC_STUB_CONST(delay_offset_ms, ());
   WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
+  WEBRTC_STUB(StartDebugRecording,
+              (const char filename[kMaxFilenameSize], int max_log_size_bytes));
   WEBRTC_STUB(StartDebugRecording, (FILE* handle));
+  WEBRTC_STUB(StartDebugRecording, (FILE * handle, int max_log_size_bytes));
   WEBRTC_STUB(StopDebugRecording, ());
   WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
   webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
@@ -1004,7 +1007,10 @@ class FakeWebRtcVoiceEngine
       float& fraction_poor_delays));
 
   WEBRTC_STUB(StartDebugRecording, (const char* fileNameUTF8));
+  WEBRTC_STUB(StartDebugRecording,
+              (const char* fileNameUTF8, int max_size_bytes));
   WEBRTC_STUB(StartDebugRecording, (FILE* handle));
+  WEBRTC_STUB(StartDebugRecording, (FILE * handle, int max_size_bytes));
   WEBRTC_STUB(StopDebugRecording, ());
 
   WEBRTC_FUNC(SetTypingDetectionStatus, (bool enable)) {