From 8267cf31ce443a88de40ef046b6f57cc3c93ef24 Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Mon, 9 Jan 2023 13:41:33 +0000
Subject: [PATCH] [Unwrap] Use RtpTimestampUnwrapper in audio_ingress
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Bug: webrtc:13982
Change-Id: I748f8e9d5497eac3335b8a9397199ddf24ecc8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288746
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39035}
---
 audio/voip/BUILD.gn         | 1 +
 audio/voip/audio_ingress.cc | 1 +
 audio/voip/audio_ingress.h  | 4 ++--
 3 files changed, 4 insertions(+), 2 deletions(-)

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