Index: webrtc/modules/audio_processing/audio_processing_unittest.cc |
diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc |
index 4d97904c6869c25febb77933fc73fbf1760ddf7d..42cf4188fcbf326f9d14fc72cc318c8817d78f04 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_unittest.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc |
@@ -680,15 +680,13 @@ void ApmTest::ProcessDelayVerificationTest(int delay_ms, int system_delay_ms, |
// limit them w.r.t. AEC delay estimation support. |
const size_t samples_per_ms = |
rtc::SafeMin<size_t>(16u, frame_->samples_per_channel_ / 10); |
- int expected_median = std::min(std::max(delay_ms - system_delay_ms, |
- delay_min), delay_max); |
- int expected_median_high = std::min( |
- std::max(expected_median + static_cast<int>(96 / samples_per_ms), |
- delay_min), |
+ const int expected_median = |
+ rtc::SafeClamp<int>(delay_ms - system_delay_ms, delay_min, delay_max); |
+ const int expected_median_high = rtc::SafeClamp<int>( |
+ expected_median + rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min, |
delay_max); |
- int expected_median_low = std::min( |
- std::max(expected_median - static_cast<int>(96 / samples_per_ms), |
- delay_min), |
+ const int expected_median_low = rtc::SafeClamp<int>( |
+ expected_median - rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min, |
delay_max); |
// Verify delay metrics. |
int median; |