Index: webrtc/modules/audio_processing/aec/aec_core.cc |
diff --git a/webrtc/modules/audio_processing/aec/aec_core.cc b/webrtc/modules/audio_processing/aec/aec_core.cc |
index e3fd14c9dab69c77eae0a807a5487cba3cb7667e..b410c55507df27e646fb1897398f7632cb4128fd 100644 |
--- a/webrtc/modules/audio_processing/aec/aec_core.cc |
+++ b/webrtc/modules/audio_processing/aec/aec_core.cc |
@@ -202,7 +202,7 @@ void BlockBuffer::Insert(const float block[PART_LEN]) { |
void BlockBuffer::ExtractExtendedBlock(float extended_block[PART_LEN2]) { |
float* block_ptr = NULL; |
- RTC_DCHECK_LT(0u, AvaliableSpace()); |
+ RTC_DCHECK_LT(0, AvaliableSpace()); |
// Extract the previous block. |
WebRtc_MoveReadPtr(buffer_, -1); |
@@ -461,7 +461,7 @@ static void UpdateLogRatioMetric(Stats* metric, float numerator, |
// Average. |
metric->counter++; |
// This is to protect overflow, which should almost never happen. |
- RTC_CHECK_NE(0u, metric->counter); |
+ RTC_CHECK_NE(0, metric->counter); |
metric->sum += metric->instant; |
metric->average = metric->sum / metric->counter; |
@@ -469,7 +469,7 @@ static void UpdateLogRatioMetric(Stats* metric, float numerator, |
if (metric->instant > metric->average) { |
metric->hicounter++; |
// This is to protect overflow, which should almost never happen. |
- RTC_CHECK_NE(0u, metric->hicounter); |
+ RTC_CHECK_NE(0, metric->hicounter); |
metric->hisum += metric->instant; |
metric->himean = metric->hisum / metric->hicounter; |
} |