diff --git a/webrtc/modules/video_coding/main/source/receiver.cc b/webrtc/modules/video_coding/main/source/receiver.cc
index e3fc0ceac7..261c51e4ef 100644
--- a/webrtc/modules/video_coding/main/source/receiver.cc
+++ b/webrtc/modules/video_coding/main/source/receiver.cc
@@ -156,12 +156,12 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(
   // Assume that render timing errors are due to changes in the video stream.
   if (next_render_time_ms < 0) {
     timing_error = true;
-  } else if (labs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
+  } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
     WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
                  VCMId(vcm_id_, receiver_id_),
                  "This frame is out of our delay bounds, resetting jitter "
                  "buffer: %d > %d",
-                 static_cast<int>(labs(next_render_time_ms - now_ms)),
+                 static_cast<int>(std::abs(next_render_time_ms - now_ms)),
                  max_video_delay_ms_);
     timing_error = true;
   } else if (static_cast<int>(timing_->TargetVideoDelay()) >
diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc
index 59f119a85d..4c0f5ed216 100644
--- a/webrtc/video/call_perf_tests.cc
+++ b/webrtc/video/call_perf_tests.cc
@@ -196,7 +196,7 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
     // estimated as being synchronized. We don't want to trigger on those.
     if (time_since_creation < kStartupTimeMs)
       return;
-    if (labs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) {
+    if (std::abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) {
       if (first_time_in_sync_ == -1) {
         first_time_in_sync_ = now_ms;
         webrtc::test::PrintResult("sync_convergence_time",