[Unwrap] Migrate NetEqDelayAnalyzer to use RtpTimestampUnwrapper

Bug: webrtc:13982
Change-Id: I35c08921c8c1be31f0de4bd81f918250bee25313
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288961
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39052}

modules/audio_coding/BUILD.gn

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 4f04b35..0bf6968 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -883,6 +883,7 @@
     "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:checks",
+    "../../rtc_base:rtc_numerics",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
     "../../rtc_base:timeutils",

modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc

diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
index 020199e..9e77457 100644
--- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
+++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
@@ -18,8 +18,8 @@
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "modules/include/module_common_types_public.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
 
 namespace webrtc {
 namespace test {
@@ -140,7 +140,7 @@
 
   std::vector<double> rtp_timestamps_ms;
   double offset = std::numeric_limits<double>::max();
-  TimestampUnwrapper unwrapper;
+  RtpTimestampUnwrapper unwrapper;
   // This loop traverses data_ and populates rtp_timestamps_ms as well as
   // calculates the base offset.
   for (auto& d : data_) {