diff --git a/audio/voip/BUILD.gn b/audio/voip/BUILD.gn
index 1b8c29e5a7..e807e2276b 100644
--- a/audio/voip/BUILD.gn
+++ b/audio/voip/BUILD.gn
@@ -70,6 +70,7 @@ rtc_library("audio_ingress") {
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../rtc_base:criticalsection",
     "../../rtc_base:logging",
+    "../../rtc_base:rtc_numerics",
     "../../rtc_base:safe_minmax",
     "../../rtc_base:timeutils",
     "../../rtc_base/synchronization:mutex",
diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index 71026e84e0..9492a51a21 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -23,6 +23,7 @@
 #include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/time_utils.h"
 
 namespace webrtc {
 
diff --git a/audio/voip/audio_ingress.h b/audio/voip/audio_ingress.h
index 9a36a46563..11bde7ce28 100644
--- a/audio/voip/audio_ingress.h
+++ b/audio/voip/audio_ingress.h
@@ -30,8 +30,8 @@
 #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/time_utils.h"
 
 namespace webrtc {
 
@@ -137,7 +137,7 @@ class AudioIngress : public AudioMixer::Source {
   // per payload type set when caller set via SetReceiveCodecs.
   std::map<int, int> receive_codec_info_ RTC_GUARDED_BY(lock_);
 
-  rtc::TimestampWrapAroundHandler timestamp_wrap_handler_ RTC_GUARDED_BY(lock_);
+  RtpTimestampUnwrapper timestamp_wrap_handler_ RTC_GUARDED_BY(lock_);
 };
 
 }  // namespace webrtc