Index: talk/media/webrtc/fakewebrtcvoiceengine.h |
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h |
index b7a7262af74ccb39e59c34b705f031406294619e..d45880763f53b5b473e4d30b16445dbd72638ba0 100644 |
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h |
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h |
@@ -112,8 +112,9 @@ class FakeAudioProcessing : public webrtc::AudioProcessing { |
WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); |
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, (FILE* handle)); |
+ WEBRTC_STUB(StartDebugRecording, |
+ (const char filename[kMaxFilenameSize], int64_t max_size_bytes)); |
+ WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes)); |
WEBRTC_STUB(StopDebugRecording, ()); |
WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); |
webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } |