From 224e39098898412d74e83deadfdef76802823c5a Mon Sep 17 00:00:00 2001
From: Evan Shrubsole <eshr@webrtc.org>
Date: Mon, 9 Jan 2023 10:22:22 +0000
Subject: [PATCH] [Unwrap] Migrate PacketArrivalHistory to
 RtpTimestampUnwrapper
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Bug: webrtc:13982
Change-Id: Idd4905c1930d51efd0b9a5a1df1ad6001f9bc37c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288941
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39037}
---
 modules/audio_coding/BUILD.gn                   |  1 +
 .../neteq/packet_arrival_history.cc             |  8 +++-----
 .../audio_coding/neteq/packet_arrival_history.h |  4 ++--
 .../neteq/packet_arrival_history_unittest.cc    | 17 +++++++++++++++++
 4 files changed, 23 insertions(+), 7 deletions(-)

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 38bff0228f..4f04b35d70 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -734,6 +734,7 @@ rtc_library("neteq") {
     "../../rtc_base:gtest_prod",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
+    "../../rtc_base:rtc_numerics",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:safe_minmax",
     "../../rtc_base:sanitizer",
diff --git a/modules/audio_coding/neteq/packet_arrival_history.cc b/modules/audio_coding/neteq/packet_arrival_history.cc
index 7196a6e393..2077383f76 100644
--- a/modules/audio_coding/neteq/packet_arrival_history.cc
+++ b/modules/audio_coding/neteq/packet_arrival_history.cc
@@ -13,7 +13,6 @@
 #include <algorithm>
 
 #include "api/neteq/tick_timer.h"
-#include "modules/include/module_common_types_public.h"
 
 namespace webrtc {
 
@@ -62,7 +61,7 @@ void PacketArrivalHistory::Reset() {
   history_.clear();
   min_packet_arrival_ = nullptr;
   max_packet_arrival_ = nullptr;
-  timestamp_unwrapper_ = TimestampUnwrapper();
+  timestamp_unwrapper_.Reset();
   newest_rtp_timestamp_ = absl::nullopt;
 }
 
@@ -70,8 +69,7 @@ int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp,
                                      int64_t time_ms) const {
   RTC_DCHECK(sample_rate_khz_ > 0);
   int64_t unwrapped_rtp_timestamp_ms =
-      timestamp_unwrapper_.UnwrapWithoutUpdate(rtp_timestamp) /
-      sample_rate_khz_;
+      timestamp_unwrapper_.PeekUnwrap(rtp_timestamp) / sample_rate_khz_;
   PacketArrival packet(unwrapped_rtp_timestamp_ms, time_ms);
   return GetPacketArrivalDelayMs(packet);
 }
@@ -88,7 +86,7 @@ bool PacketArrivalHistory::IsNewestRtpTimestamp(uint32_t rtp_timestamp) const {
     return false;
   }
   int64_t unwrapped_rtp_timestamp =
-      timestamp_unwrapper_.UnwrapWithoutUpdate(rtp_timestamp);
+      timestamp_unwrapper_.PeekUnwrap(rtp_timestamp);
   return unwrapped_rtp_timestamp == *newest_rtp_timestamp_;
 }
 
diff --git a/modules/audio_coding/neteq/packet_arrival_history.h b/modules/audio_coding/neteq/packet_arrival_history.h
index 79fc9176bc..cad362b469 100644
--- a/modules/audio_coding/neteq/packet_arrival_history.h
+++ b/modules/audio_coding/neteq/packet_arrival_history.h
@@ -16,7 +16,7 @@
 
 #include "absl/types/optional.h"
 #include "api/neteq/tick_timer.h"
-#include "modules/include/module_common_types_public.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
 
 namespace webrtc {
 
@@ -72,7 +72,7 @@ class PacketArrivalHistory {
   const PacketArrival* min_packet_arrival_ = nullptr;
   const PacketArrival* max_packet_arrival_ = nullptr;
   const int window_size_ms_;
-  TimestampUnwrapper timestamp_unwrapper_;
+  RtpTimestampUnwrapper timestamp_unwrapper_;
   absl::optional<int64_t> newest_rtp_timestamp_;
   int sample_rate_khz_ = 0;
 };
diff --git a/modules/audio_coding/neteq/packet_arrival_history_unittest.cc b/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
index 286a7acb2c..539a318fe1 100644
--- a/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
+++ b/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
@@ -120,5 +120,22 @@ TEST_F(PacketArrivalHistoryTest, TimestampWraparound) {
   EXPECT_EQ(history_.GetMaxDelayMs(), 3 * kFrameSizeMs);
 }
 
+TEST_F(PacketArrivalHistoryTest, TimestampWraparoundBackwards) {
+  timestamp_ = 0;
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  // Insert timestamp that will wrap around.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), kFrameSizeMs);
+
+  // Insert reordered packet before the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(-2 * kFrameSizeMs), 3 * kFrameSizeMs);
+
+  // Insert another in-order packet after the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
+
+  EXPECT_EQ(history_.GetMaxDelayMs(), 3 * kFrameSizeMs);
+}
+
 }  // namespace
 }  // namespace webrtc