Index: talk/media/webrtc/fakewebrtcvoiceengine.h |
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h |
index 1167b6b9830c2f9edd1ff0b9caa606a50814d372..91d39eb9a14ef633439a8bd22a08fc6c91b28af5 100644 |
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h |
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h |
@@ -156,8 +156,9 @@ 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; } |