From 2412602b6801ae7495c7506fed47d573660c3545 Mon Sep 17 00:00:00 2001
From: Tim Na <natim@webrtc.org>
Date: Fri, 11 Dec 2020 13:31:55 -0800
Subject: [PATCH] Using absl::optional for round trip time return type
 handling.

No-Try: True
Bug: webrtc:11989
Change-Id: If2ed9b83468c03b82b372e64d8012e5786295476
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197060
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32827}
---
 audio/voip/BUILD.gn         |  1 +
 audio/voip/audio_ingress.cc | 16 ++++++++++------
 audio/voip/audio_ingress.h  |  9 +++++----
 3 files changed, 16 insertions(+), 10 deletions(-)

diff --git a/audio/voip/BUILD.gn b/audio/voip/BUILD.gn
index f4b6142a74..dd5267f55a 100644
--- a/audio/voip/BUILD.gn
+++ b/audio/voip/BUILD.gn
@@ -78,6 +78,7 @@ rtc_library("audio_ingress") {
     "../../rtc_base/synchronization:mutex",
     "../utility:audio_frame_operations",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("audio_egress") {
diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index 07def99559..3be471824d 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -184,8 +184,8 @@ void AudioIngress::ReceivedRTCPPacket(
   // Deliver RTCP packet to RTP/RTCP module for parsing.
   rtp_rtcp_->IncomingRtcpPacket(rtcp_packet.data(), rtcp_packet.size());
 
-  int64_t rtt = GetRoundTripTime();
-  if (rtt == -1) {
+  absl::optional<int64_t> rtt = GetRoundTripTime();
+  if (!rtt.has_value()) {
     // Waiting for valid RTT.
     return;
   }
@@ -199,18 +199,18 @@ void AudioIngress::ReceivedRTCPPacket(
 
   {
     MutexLock lock(&lock_);
-    ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
+    ntp_estimator_.UpdateRtcpTimestamp(*rtt, ntp_secs, ntp_frac, rtp_timestamp);
   }
 }
 
-int64_t AudioIngress::GetRoundTripTime() {
+absl::optional<int64_t> AudioIngress::GetRoundTripTime() {
   const std::vector<ReportBlockData>& report_data =
       rtp_rtcp_->GetLatestReportBlockData();
 
   // If we do not have report block which means remote RTCP hasn't be received
   // yet, return -1 as to indicate uninitialized value.
   if (report_data.empty()) {
-    return -1;
+    return absl::nullopt;
   }
 
   // We don't know in advance the remote SSRC used by the other end's receiver
@@ -226,7 +226,11 @@ int64_t AudioIngress::GetRoundTripTime() {
     rtp_rtcp_->SetRemoteSSRC(sender_ssrc);
   }
 
-  return (block_data.has_rtt() ? block_data.last_rtt_ms() : -1);
+  if (!block_data.has_rtt()) {
+    return absl::nullopt;
+  }
+
+  return block_data.last_rtt_ms();
 }
 
 }  // namespace webrtc
diff --git a/audio/voip/audio_ingress.h b/audio/voip/audio_ingress.h
index d3680e0f00..663b59bc67 100644
--- a/audio/voip/audio_ingress.h
+++ b/audio/voip/audio_ingress.h
@@ -17,6 +17,7 @@
 #include <memory>
 #include <utility>
 
+#include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/audio/audio_mixer.h"
 #include "api/rtp_headers.h"
@@ -78,10 +79,6 @@ class AudioIngress : public AudioMixer::Source {
     return output_audio_level_.TotalDuration();
   }
 
-  // Returns network round trip time (RTT) measued by RTCP exchange with
-  // remote media endpoint. RTT value -1 indicates that it's not initialized.
-  int64_t GetRoundTripTime();
-
   NetworkStatistics GetNetworkStatistics() const {
     NetworkStatistics stats;
     acm_receiver_.GetNetworkStatistics(&stats,
@@ -105,6 +102,10 @@ class AudioIngress : public AudioMixer::Source {
   }
 
  private:
+  // Returns network round trip time (RTT) measued by RTCP exchange with
+  // remote media endpoint. Returns absl::nullopt when it's not initialized.
+  absl::optional<int64_t> GetRoundTripTime();
+
   // Indicates AudioIngress status as caller invokes Start/StopPlaying.
   // If not playing, incoming RTP data processing is skipped, thus
   // producing no data to output device.