diff --git a/aegisub/audio_provider_lavc.cpp b/aegisub/audio_provider_lavc.cpp
index a46941e43..5a128e3dc 100644
--- a/aegisub/audio_provider_lavc.cpp
+++ b/aegisub/audio_provider_lavc.cpp
@@ -197,14 +197,14 @@ void LAVCAudioProvider::GetAudio(void *buf, int64_t start, int64_t count)
 					decoded_samples = audio_resample(rsct, _buf, buffer, decoded_samples / codecContext->channels);
 
 					/* make some noise if we somehow ended up with more samples than we wanted (will cause audio skew) */
-					if (decoded_samples <= samples_to_decode)
+					if (decoded_samples > samples_to_decode)
 						wxLogMessage(wxString::Format(_T("Warning: decoder output more samples than requested, audio skew highly likely! (Wanted %d, got %d)"), samples_to_decode, decoded_samples));
 				} else {
 					/* no resampling needed, just copy to the buffer */
 					/* if (decoded_samples > samples_to_decode)
 						decoded_samples = samples_to_decode; */
 					/* I do not understand the point of the above -Fluff */
-					if (decoded_samples <= samples_to_decode)
+					if (decoded_samples > samples_to_decode)
 						wxLogMessage(wxString::Format(_T("Warning: decoder output more samples than requested, audio skew highly likely! (Wanted %d, got %d)"), samples_to_decode, decoded_samples));
 
 					memcpy(_buf, buffer, temp_output_buffer_size);