Replace rtc::Optional with absl::optional

This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}

modules/congestion_controller/bbr/BUILD.gn

@@ -31,12 +31,12 @@ rtc_source_set("bbr_controller") {
     ":loss_rate_filter",
     ":rtt_stats",
     ":windowed_filter",
-    "../../../api:optional",
     "../../../api/transport:network_control",
     "../../../rtc_base:checks",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/experiments:congestion_controller_experiment",
     "../../../rtc_base/system:fallthrough",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 
@@ -48,13 +48,13 @@ rtc_source_set("bandwidth_sampler") {
   ]
   deps = [
     ":packet_number_indexed_queue",
-    "../../../api:optional",
     "../../../api/units:data_rate",
     "../../../api/units:data_size",
     "../../../api/units:time_delta",
     "../../../api/units:timestamp",
     "../../../rtc_base:checks",
     "../../../rtc_base:rtc_base_approved",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 
@@ -90,7 +90,7 @@ rtc_source_set("loss_rate_filter") {
     "loss_rate_filter.h",
   ]
   deps = [
-    "../../../api:optional",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 rtc_source_set("rtt_stats") {

modules/congestion_controller/bbr/bandwidth_sampler.h

@@ -12,7 +12,7 @@
 #ifndef MODULES_CONGESTION_CONTROLLER_BBR_BANDWIDTH_SAMPLER_H_
 #define MODULES_CONGESTION_CONTROLLER_BBR_BANDWIDTH_SAMPLER_H_
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "api/units/data_rate.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
@@ -187,11 +187,11 @@ class BandwidthSampler {
 
     // The value of |last_acked_packet_sent_time_| at the time the packet was
     // sent.
-    rtc::Optional<Timestamp> last_acked_packet_sent_time;
+    absl::optional<Timestamp> last_acked_packet_sent_time;
 
     // The value of |last_acked_packet_ack_time_| at the time the packet was
     // sent.
-    rtc::Optional<Timestamp> last_acked_packet_ack_time;
+    absl::optional<Timestamp> last_acked_packet_ack_time;
 
     // The value of |total_data_acked_| at the time the packet was
     // sent.
@@ -227,10 +227,10 @@ class BandwidthSampler {
 
   // The time at which the last acknowledged packet was sent. Set to
   // Timestamp::Zero() if no valid timestamp is available.
-  rtc::Optional<Timestamp> last_acked_packet_sent_time_;
+  absl::optional<Timestamp> last_acked_packet_sent_time_;
 
   // The time at which the most recent packet was acknowledged.
-  rtc::Optional<Timestamp> last_acked_packet_ack_time_;
+  absl::optional<Timestamp> last_acked_packet_ack_time_;
 
   // The most recently sent packet.
   int64_t last_sent_packet_;

modules/congestion_controller/bbr/bbr_network_controller.cc

@@ -382,7 +382,7 @@ bool BbrNetworkController::IsProbingForMoreBandwidth() const {
 NetworkControlUpdate BbrNetworkController::OnTransportPacketsFeedback(
     TransportPacketsFeedback msg) {
   Timestamp feedback_recv_time = msg.feedback_time;
-  rtc::Optional<SentPacket> last_sent_packet =
+  absl::optional<SentPacket> last_sent_packet =
       msg.PacketsWithFeedback().back().sent_packet;
   if (!last_sent_packet.has_value()) {
     RTC_LOG(LS_WARNING) << "Last ack packet not in history, no RTT update";

modules/congestion_controller/bbr/bbr_network_controller.h

@@ -26,7 +26,7 @@
 #include "modules/congestion_controller/bbr/rtt_stats.h"
 #include "modules/congestion_controller/bbr/windowed_filter.h"
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "rtc_base/random.h"
 
 namespace webrtc {
@@ -217,7 +217,7 @@ class BbrNetworkController : public NetworkControllerInterface {
   webrtc::Random random_;
   LossRateFilter loss_rate_;
 
-  rtc::Optional<TargetRateConstraints> constraints_;
+  absl::optional<TargetRateConstraints> constraints_;
 
   Mode mode_;
 
@@ -244,7 +244,7 @@ class BbrNetworkController : public NetworkControllerInterface {
   MaxAckHeightFilter max_ack_height_;
 
   // The time this aggregation started and the number of bytes acked during it.
-  rtc::Optional<Timestamp> aggregation_epoch_start_time_;
+  absl::optional<Timestamp> aggregation_epoch_start_time_;
   DataSize aggregation_epoch_bytes_;
 
   // The number of bytes acknowledged since the last time bytes in flight
@@ -308,7 +308,7 @@ class BbrNetworkController : public NetworkControllerInterface {
   // Time at which PROBE_RTT has to be exited.  Setting it to zero indicates
   // that the time is yet unknown as the number of packets in flight has not
   // reached the required value.
-  rtc::Optional<Timestamp> exit_probe_rtt_at_;
+  absl::optional<Timestamp> exit_probe_rtt_at_;
   // Indicates whether a round-trip has passed since PROBE_RTT became active.
   bool probe_rtt_round_passed_;
 
@@ -321,7 +321,7 @@ class BbrNetworkController : public NetworkControllerInterface {
   // Receiving acknowledgement of a packet after |end_recovery_at_| will cause
   // BBR to exit the recovery mode.  A set value indicates at least one
   // loss has been detected, so it must not be reset.
-  rtc::Optional<int64_t> end_recovery_at_;
+  absl::optional<int64_t> end_recovery_at_;
   // A window used to limit the number of bytes in flight during loss recovery.
   DataSize recovery_window_;
 

modules/congestion_controller/bbr/loss_rate_filter.h

@@ -10,7 +10,7 @@
 #ifndef MODULES_CONGESTION_CONTROLLER_BBR_LOSS_RATE_FILTER_H_
 #define MODULES_CONGESTION_CONTROLLER_BBR_LOSS_RATE_FILTER_H_
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 
 namespace webrtc {
 namespace bbr {

modules/congestion_controller/delay_based_bwe.cc

@@ -114,7 +114,7 @@ DelayBasedBwe::~DelayBasedBwe() {}
 
 DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
     const std::vector<PacketFeedback>& packet_feedback_vector,
-    rtc::Optional<uint32_t> acked_bitrate_bps) {
+    absl::optional<uint32_t> acked_bitrate_bps) {
   RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
                             packet_feedback_vector.end(),
                             PacketFeedbackComparator()));
@@ -222,12 +222,12 @@ void DelayBasedBwe::IncomingPacketFeedback(
 }
 
 DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
-    rtc::Optional<uint32_t> acked_bitrate_bps,
+    absl::optional<uint32_t> acked_bitrate_bps,
     bool recovered_from_overuse) {
   Result result;
   int64_t now_ms = clock_->TimeInMilliseconds();
 
-  rtc::Optional<int> probe_bitrate_bps =
+  absl::optional<int> probe_bitrate_bps =
       probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
   // Currently overusing the bandwidth.
   if (delay_detector_->State() == BandwidthUsage::kBwOverusing) {
@@ -280,7 +280,7 @@ DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
 }
 
 bool DelayBasedBwe::UpdateEstimate(int64_t now_ms,
-                                   rtc::Optional<uint32_t> acked_bitrate_bps,
+                                   absl::optional<uint32_t> acked_bitrate_bps,
                                    uint32_t* target_bitrate_bps) {
   // TODO(terelius): RateControlInput::noise_var is deprecated and will be
   // removed. In the meantime, we set it to zero.

modules/congestion_controller/delay_based_bwe.h

@@ -47,7 +47,7 @@ class DelayBasedBwe {
 
   Result IncomingPacketFeedbackVector(
       const std::vector<PacketFeedback>& packet_feedback_vector,
-      rtc::Optional<uint32_t> acked_bitrate_bps);
+      absl::optional<uint32_t> acked_bitrate_bps);
   void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms);
   bool LatestEstimate(std::vector<uint32_t>* ssrcs,
                       uint32_t* bitrate_bps) const;
@@ -58,12 +58,12 @@ class DelayBasedBwe {
  private:
   void IncomingPacketFeedback(const PacketFeedback& packet_feedback);
   Result OnLongFeedbackDelay(int64_t arrival_time_ms);
-  Result MaybeUpdateEstimate(rtc::Optional<uint32_t> acked_bitrate_bps,
+  Result MaybeUpdateEstimate(absl::optional<uint32_t> acked_bitrate_bps,
                              bool request_probe);
   // Updates the current remote rate estimate and returns true if a valid
   // estimate exists.
   bool UpdateEstimate(int64_t now_ms,
-                      rtc::Optional<uint32_t> acked_bitrate_bps,
+                      absl::optional<uint32_t> acked_bitrate_bps,
                       uint32_t* target_bitrate_bps);
 
   rtc::RaceChecker network_race_;

modules/congestion_controller/delay_based_bwe_unittest.cc

@@ -29,7 +29,7 @@ constexpr float kTargetUtilizationFraction = 0.95f;
 TEST_F(LegacyDelayBasedBweTest, NoCrashEmptyFeedback) {
   std::vector<PacketFeedback> packet_feedback_vector;
   bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector,
-                                                   rtc::nullopt);
+                                                   absl::nullopt);
 }
 
 TEST_F(LegacyDelayBasedBweTest, NoCrashOnlyLostFeedback) {
@@ -41,7 +41,7 @@ TEST_F(LegacyDelayBasedBweTest, NoCrashOnlyLostFeedback) {
                                                   PacketFeedback::kNoSendTime,
                                                   1, 1500, PacedPacketInfo()));
   bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector,
-                                                   rtc::nullopt);
+                                                   absl::nullopt);
 }
 
 TEST_F(LegacyDelayBasedBweTest, ProbeDetection) {

modules/congestion_controller/goog_cc/BUILD.gn

@@ -43,7 +43,6 @@ rtc_static_library("goog_cc") {
     "../..:module_api",
     "../../..:webrtc_common",
     "../../../:typedefs",
-    "../../../api:optional",
     "../../../api/transport:network_control",
     "../../../logging:rtc_event_log_api",
     "../../../logging:rtc_event_pacing",
@@ -57,6 +56,7 @@ rtc_static_library("goog_cc") {
     "../../pacing",
     "../../remote_bitrate_estimator",
     "../../rtp_rtcp:rtp_rtcp_format",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 
@@ -85,7 +85,6 @@ rtc_source_set("estimators") {
   }
 
   deps = [
-    "../../../api:optional",
     "../../../logging:rtc_event_bwe",
     "../../../logging:rtc_event_log_api",
     "../../../rtc_base:checks",
@@ -96,6 +95,7 @@ rtc_source_set("estimators") {
     "../../../system_wrappers:metrics_api",
     "../../remote_bitrate_estimator",
     "../../rtp_rtcp:rtp_rtcp_format",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 

modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc

@@ -47,7 +47,7 @@ void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
   }
 }
 
-rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
+absl::optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
   return bitrate_estimator_->bitrate_bps();
 }
 

modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h

@@ -14,7 +14,7 @@
 #include <memory>
 #include <vector>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
 
 namespace webrtc {
@@ -31,12 +31,12 @@ class AcknowledgedBitrateEstimator {
 
   void IncomingPacketFeedbackVector(
       const std::vector<PacketFeedback>& packet_feedback_vector);
-  rtc::Optional<uint32_t> bitrate_bps() const;
+  absl::optional<uint32_t> bitrate_bps() const;
   void SetAlrEndedTimeMs(int64_t alr_ended_time_ms);
 
  private:
   void MaybeExpectFastRateChange(int64_t packet_arrival_time_ms);
-  rtc::Optional<int64_t> alr_ended_time_ms_;
+  absl::optional<int64_t> alr_ended_time_ms_;
   std::unique_ptr<BitrateEstimator> bitrate_estimator_;
 };
 

modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc

@@ -35,7 +35,7 @@ constexpr size_t kPayloadSize = 10;
 class MockBitrateEstimator : public BitrateEstimator {
  public:
   MOCK_METHOD2(Update, void(int64_t now_ms, int bytes));
-  MOCK_CONST_METHOD0(bitrate_bps, rtc::Optional<uint32_t>());
+  MOCK_CONST_METHOD0(bitrate_bps, absl::optional<uint32_t>());
   MOCK_METHOD0(ExpectFastRateChange, void());
 };
 
@@ -124,7 +124,7 @@ TEST(TestAcknowledgedBitrateEstimator, ExpectFastRateChangeWhenLeftAlr) {
 
 TEST(TestAcknowledgedBitrateEstimator, ReturnBitrate) {
   auto states = CreateTestStates();
-  rtc::Optional<uint32_t> return_value(42);
+  absl::optional<uint32_t> return_value(42);
   EXPECT_CALL(*states.mock_bitrate_estimator, bitrate_bps())
       .Times(1)
       .WillOnce(Return(return_value));

modules/congestion_controller/goog_cc/alr_detector.cc

@@ -35,7 +35,7 @@ AlrDetector::AlrDetector(RtcEventLog* event_log)
       alr_budget_(0, true),
       event_log_(event_log) {
   RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
-  rtc::Optional<AlrExperimentSettings> experiment_settings =
+  absl::optional<AlrExperimentSettings> experiment_settings =
       AlrExperimentSettings::CreateFromFieldTrial(
           AlrExperimentSettings::kScreenshareProbingBweExperimentName);
   if (!experiment_settings) {
@@ -89,7 +89,7 @@ void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
   alr_budget_.set_target_rate_kbps(rtc::dchecked_cast<int>(target_rate_kbps));
 }
 
-rtc::Optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
+absl::optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
     const {
   return alr_started_time_ms_;
 }

modules/congestion_controller/goog_cc/alr_detector.h

@@ -11,7 +11,7 @@
 #ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
 #define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/pacing/interval_budget.h"
 #include "modules/pacing/paced_sender.h"
@@ -44,7 +44,7 @@ class AlrDetector {
 
   // Returns time in milliseconds when the current application-limited region
   // started or empty result if the sender is currently not application-limited.
-  rtc::Optional<int64_t> GetApplicationLimitedRegionStartTime() const;
+  absl::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
 
   // Sent traffic percentage as a function of network capacity used to determine
   // application-limited region. ALR region start when bandwidth usage drops
@@ -63,10 +63,10 @@ class AlrDetector {
   int alr_start_budget_level_percent_;
   int alr_stop_budget_level_percent_;
 
-  rtc::Optional<int64_t> last_send_time_ms_;
+  absl::optional<int64_t> last_send_time_ms_;
 
   IntervalBudget alr_budget_;
-  rtc::Optional<int64_t> alr_started_time_ms_;
+  absl::optional<int64_t> alr_started_time_ms_;
 
   RtcEventLog* event_log_;
 };

modules/congestion_controller/goog_cc/alr_detector_unittest.cc

@@ -65,8 +65,8 @@ class SimulateOutgoingTrafficIn {
   }
   AlrDetector* const alr_detector_;
   int64_t* timestamp_ms_;
-  rtc::Optional<int> interval_ms_;
+  absl::optional<int> interval_ms_;
-  rtc::Optional<int> usage_percentage_;
+  absl::optional<int> usage_percentage_;
 };
 }  // namespace
 
@@ -152,7 +152,7 @@ TEST_F(AlrDetectorTest, BandwidthEstimateChanges) {
 TEST_F(AlrDetectorTest, ParseControlFieldTrial) {
   webrtc::test::ScopedFieldTrials field_trial(
       "WebRTC-ProbingScreenshareBwe/Control/");
-  rtc::Optional<AlrExperimentSettings> parsed_params =
+  absl::optional<AlrExperimentSettings> parsed_params =
       AlrExperimentSettings::CreateFromFieldTrial(
           "WebRTC-ProbingScreenshareBwe");
   EXPECT_FALSE(static_cast<bool>(parsed_params));
@@ -161,7 +161,7 @@ TEST_F(AlrDetectorTest, ParseControlFieldTrial) {
 TEST_F(AlrDetectorTest, ParseActiveFieldTrial) {
   webrtc::test::ScopedFieldTrials field_trial(
       "WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/");
-  rtc::Optional<AlrExperimentSettings> parsed_params =
+  absl::optional<AlrExperimentSettings> parsed_params =
       AlrExperimentSettings::CreateFromFieldTrial(
           "WebRTC-ProbingScreenshareBwe");
   ASSERT_TRUE(static_cast<bool>(parsed_params));

modules/congestion_controller/goog_cc/bitrate_estimator.cc

@@ -92,9 +92,9 @@ float BitrateEstimator::UpdateWindow(int64_t now_ms,
   return bitrate_sample;
 }
 
-rtc::Optional<uint32_t> BitrateEstimator::bitrate_bps() const {
+absl::optional<uint32_t> BitrateEstimator::bitrate_bps() const {
   if (bitrate_estimate_ < 0.f)
-    return rtc::nullopt;
+    return absl::nullopt;
   return bitrate_estimate_ * 1000;
 }
 

modules/congestion_controller/goog_cc/bitrate_estimator.h

@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 
 namespace webrtc {
 
@@ -28,7 +28,7 @@ class BitrateEstimator {
   virtual ~BitrateEstimator();
   virtual void Update(int64_t now_ms, int bytes);
 
-  virtual rtc::Optional<uint32_t> bitrate_bps() const;
+  virtual absl::optional<uint32_t> bitrate_bps() const;
 
   virtual void ExpectFastRateChange();
 

modules/congestion_controller/goog_cc/delay_based_bwe.cc

@@ -114,7 +114,7 @@ DelayBasedBwe::~DelayBasedBwe() {}
 
 DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
     const std::vector<PacketFeedback>& packet_feedback_vector,
-    rtc::Optional<uint32_t> acked_bitrate_bps,
+    absl::optional<uint32_t> acked_bitrate_bps,
     int64_t at_time_ms) {
   RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
                             packet_feedback_vector.end(),
@@ -225,13 +225,13 @@ void DelayBasedBwe::IncomingPacketFeedback(
 }
 
 DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
-    rtc::Optional<uint32_t> acked_bitrate_bps,
+    absl::optional<uint32_t> acked_bitrate_bps,
     bool recovered_from_overuse,
     int64_t at_time_ms) {
   Result result;
   int64_t now_ms = at_time_ms;
 
-  rtc::Optional<int> probe_bitrate_bps =
+  absl::optional<int> probe_bitrate_bps =
       probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
   // Currently overusing the bandwidth.
   if (delay_detector_->State() == BandwidthUsage::kBwOverusing) {
@@ -284,7 +284,7 @@ DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
 }
 
 bool DelayBasedBwe::UpdateEstimate(int64_t now_ms,
-                                   rtc::Optional<uint32_t> acked_bitrate_bps,
+                                   absl::optional<uint32_t> acked_bitrate_bps,
                                    uint32_t* target_bitrate_bps) {
   // TODO(terelius): RateControlInput::noise_var is deprecated and will be
   // removed. In the meantime, we set it to zero.

modules/congestion_controller/goog_cc/delay_based_bwe.h

@@ -48,7 +48,7 @@ class DelayBasedBwe {
 
   Result IncomingPacketFeedbackVector(
       const std::vector<PacketFeedback>& packet_feedback_vector,
-      rtc::Optional<uint32_t> acked_bitrate_bps,
+      absl::optional<uint32_t> acked_bitrate_bps,
       int64_t at_time_ms);
   void OnRttUpdate(int64_t avg_rtt_ms);
   bool LatestEstimate(std::vector<uint32_t>* ssrcs,
@@ -61,13 +61,13 @@ class DelayBasedBwe {
   void IncomingPacketFeedback(const PacketFeedback& packet_feedback,
                               int64_t at_time_ms);
   Result OnLongFeedbackDelay(int64_t arrival_time_ms);
-  Result MaybeUpdateEstimate(rtc::Optional<uint32_t> acked_bitrate_bps,
+  Result MaybeUpdateEstimate(absl::optional<uint32_t> acked_bitrate_bps,
                              bool request_probe,
                              int64_t at_time_ms);
   // Updates the current remote rate estimate and returns true if a valid
   // estimate exists.
   bool UpdateEstimate(int64_t now_ms,
-                      rtc::Optional<uint32_t> acked_bitrate_bps,
+                      absl::optional<uint32_t> acked_bitrate_bps,
                       uint32_t* target_bitrate_bps);
 
   rtc::RaceChecker network_race_;

modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc

@@ -31,7 +31,7 @@ constexpr int64_t kDummyTimestamp = 1000;
 TEST_F(DelayBasedBweTest, NoCrashEmptyFeedback) {
   std::vector<PacketFeedback> packet_feedback_vector;
   bitrate_estimator_->IncomingPacketFeedbackVector(
-      packet_feedback_vector, rtc::nullopt, kDummyTimestamp);
+      packet_feedback_vector, absl::nullopt, kDummyTimestamp);
 }
 
 TEST_F(DelayBasedBweTest, NoCrashOnlyLostFeedback) {
@@ -43,7 +43,7 @@ TEST_F(DelayBasedBweTest, NoCrashOnlyLostFeedback) {
                                                   PacketFeedback::kNoSendTime,
                                                   1, 1500, PacedPacketInfo()));
   bitrate_estimator_->IncomingPacketFeedbackVector(
-      packet_feedback_vector, rtc::nullopt, kDummyTimestamp);
+      packet_feedback_vector, absl::nullopt, kDummyTimestamp);
 }
 
 TEST_F(DelayBasedBweTest, ProbeDetection) {

modules/congestion_controller/goog_cc/goog_cc_network_control.cc

@@ -101,7 +101,7 @@ std::vector<PacketFeedback> ReceivedPacketsFeedbackAsRtp(
   return packet_feedback_vector;
 }
 
-int64_t GetBpsOrDefault(const rtc::Optional<DataRate>& rate,
+int64_t GetBpsOrDefault(const absl::optional<DataRate>& rate,
                         int64_t fallback_bps) {
   if (rate && rate->IsFinite()) {
     return rate->bps();
@@ -175,7 +175,7 @@ NetworkControlUpdate GoogCcNetworkController::OnNetworkRouteChange(
 NetworkControlUpdate GoogCcNetworkController::OnProcessInterval(
     ProcessInterval msg) {
   bandwidth_estimation_->UpdateEstimate(msg.at_time.ms());
-  rtc::Optional<int64_t> start_time_ms =
+  absl::optional<int64_t> start_time_ms =
       alr_detector_->GetApplicationLimitedRegionStartTime();
   probe_controller_->SetAlrStartTimeMs(start_time_ms);
   probe_controller_->Process(msg.at_time.ms());
@@ -244,13 +244,13 @@ NetworkControlUpdate GoogCcNetworkController::OnStreamsConfig(
 
 NetworkControlUpdate GoogCcNetworkController::OnTargetRateConstraints(
     TargetRateConstraints constraints) {
-  UpdateBitrateConstraints(constraints, rtc::nullopt);
+  UpdateBitrateConstraints(constraints, absl::nullopt);
   return MaybeTriggerOnNetworkChanged(constraints.at_time);
 }
 
 void GoogCcNetworkController::UpdateBitrateConstraints(
     TargetRateConstraints constraints,
-    rtc::Optional<DataRate> starting_rate) {
+    absl::optional<DataRate> starting_rate) {
   int64_t min_bitrate_bps = GetBpsOrDefault(constraints.min_data_rate, 0);
   int64_t max_bitrate_bps = GetBpsOrDefault(constraints.max_data_rate, -1);
   int64_t start_bitrate_bps = GetBpsOrDefault(starting_rate, -1);
@@ -301,7 +301,7 @@ NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
   std::vector<PacketFeedback> received_feedback_vector =
       ReceivedPacketsFeedbackAsRtp(report);
 
-  rtc::Optional<int64_t> alr_start_time =
+  absl::optional<int64_t> alr_start_time =
       alr_detector_->GetApplicationLimitedRegionStartTime();
 
   if (previously_in_alr && !alr_start_time.has_value()) {
@@ -336,13 +336,14 @@ NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
   return update;
 }
 
-rtc::Optional<DataSize> GoogCcNetworkController::MaybeUpdateCongestionWindow() {
+absl::optional<DataSize>
+GoogCcNetworkController::MaybeUpdateCongestionWindow() {
   if (!in_cwnd_experiment_)
-    return rtc::nullopt;
+    return absl::nullopt;
   // No valid RTT. Could be because send-side BWE isn't used, in which case
   // we don't try to limit the outstanding packets.
   if (!min_feedback_rtt_ms_)
-    return rtc::nullopt;
+    return absl::nullopt;
 
   const DataSize kMinCwnd = DataSize::bytes(2 * 1500);
   TimeDelta time_window =

modules/congestion_controller/goog_cc/goog_cc_network_control.h

@@ -16,7 +16,7 @@
 #include <memory>
 #include <vector>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "api/transport/network_control.h"
 #include "logging/rtc_event_log/rtc_event_log.h"
 #include "modules/bitrate_controller/send_side_bandwidth_estimation.h"
@@ -51,8 +51,8 @@ class GoogCcNetworkController : public NetworkControllerInterface {
 
  private:
   void UpdateBitrateConstraints(TargetRateConstraints constraints,
-                                rtc::Optional<DataRate> starting_rate);
+                                absl::optional<DataRate> starting_rate);
-  rtc::Optional<DataSize> MaybeUpdateCongestionWindow();
+  absl::optional<DataSize> MaybeUpdateCongestionWindow();
   NetworkControlUpdate MaybeTriggerOnNetworkChanged(Timestamp at_time);
   bool GetNetworkParameters(int32_t* estimated_bitrate_bps,
                             uint8_t* fraction_loss,
@@ -71,10 +71,10 @@ class GoogCcNetworkController : public NetworkControllerInterface {
   std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator_;
 
   std::deque<int64_t> feedback_rtts_;
-  rtc::Optional<int64_t> min_feedback_rtt_ms_;
+  absl::optional<int64_t> min_feedback_rtt_ms_;
 
   DataRate last_bandwidth_;
-  rtc::Optional<TargetTransferRate> last_target_rate_;
+  absl::optional<TargetTransferRate> last_target_rate_;
 
   int32_t last_estimated_bitrate_bps_ = 0;
   uint8_t last_estimated_fraction_loss_ = 0;
@@ -89,7 +89,7 @@ class GoogCcNetworkController : public NetworkControllerInterface {
   int64_t accepted_queue_ms_;
   bool previously_in_alr = false;
 
-  rtc::Optional<DataSize> current_data_window_;
+  absl::optional<DataSize> current_data_window_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GoogCcNetworkController);
 };

modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc

@@ -170,9 +170,9 @@ int ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
   return *estimated_bitrate_bps_;
 }
 
-rtc::Optional<int>
+absl::optional<int>
 ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrateBps() {
-  rtc::Optional<int> estimated_bitrate_bps = estimated_bitrate_bps_;
+  absl::optional<int> estimated_bitrate_bps = estimated_bitrate_bps_;
   estimated_bitrate_bps_.reset();
   return estimated_bitrate_bps;
 }

modules/congestion_controller/goog_cc/probe_bitrate_estimator.h

@@ -28,7 +28,7 @@ class ProbeBitrateEstimator {
   // Returns the estimated bitrate if the probe completes a valid cluster.
   int HandleProbeAndEstimateBitrate(const PacketFeedback& packet_feedback);
 
-  rtc::Optional<int> FetchAndResetLastEstimatedBitrateBps();
+  absl::optional<int> FetchAndResetLastEstimatedBitrateBps();
 
  private:
   struct AggregatedCluster {
@@ -47,7 +47,7 @@ class ProbeBitrateEstimator {
 
   std::map<int, AggregatedCluster> clusters_;
   RtcEventLog* const event_log_;
-  rtc::Optional<int> estimated_bitrate_bps_;
+  absl::optional<int> estimated_bitrate_bps_;
 };
 
 }  // namespace webrtc

modules/congestion_controller/goog_cc/probe_controller.cc

@@ -209,7 +209,7 @@ void ProbeController::EnablePeriodicAlrProbing(bool enable) {
 }
 
 void ProbeController::SetAlrStartTimeMs(
-    rtc::Optional<int64_t> alr_start_time_ms) {
+    absl::optional<int64_t> alr_start_time_ms) {
   alr_start_time_ms_ = alr_start_time_ms;
 }
 void ProbeController::SetAlrEndedTimeMs(int64_t alr_end_time_ms) {

modules/congestion_controller/goog_cc/probe_controller.h

@@ -16,7 +16,7 @@
 #include <initializer_list>
 #include <vector>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "api/transport/network_control.h"
 #include "rtc_base/constructormagic.h"
 
@@ -50,7 +50,7 @@ class ProbeController {
 
   void EnablePeriodicAlrProbing(bool enable);
 
-  void SetAlrStartTimeMs(rtc::Optional<int64_t> alr_start_time);
+  void SetAlrStartTimeMs(absl::optional<int64_t> alr_start_time);
   void SetAlrEndedTimeMs(int64_t alr_end_time);
 
   void RequestProbe(int64_t at_time_ms);
@@ -86,8 +86,8 @@ class ProbeController {
   int64_t start_bitrate_bps_;
   int64_t max_bitrate_bps_;
   int64_t last_bwe_drop_probing_time_ms_;
-  rtc::Optional<int64_t> alr_start_time_ms_;
+  absl::optional<int64_t> alr_start_time_ms_;
-  rtc::Optional<int64_t> alr_end_time_ms_;
+  absl::optional<int64_t> alr_end_time_ms_;
   bool enable_periodic_alr_probing_;
   int64_t time_of_last_large_drop_ms_;
   int64_t bitrate_before_last_large_drop_bps_;

modules/congestion_controller/goog_cc/probe_controller_unittest.cc

@@ -161,7 +161,7 @@ TEST_F(ProbeControllerTest, RequestProbeWhenAlrEndedRecently) {
   probe_controller_->SetEstimatedBitrate(500, NowMs());
   EXPECT_EQ(probe_controller_->GetAndResetPendingProbes().size(), 2u);
 
-  probe_controller_->SetAlrStartTimeMs(rtc::nullopt);
+  probe_controller_->SetAlrStartTimeMs(absl::nullopt);
   clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
   probe_controller_->Process(NowMs());
   probe_controller_->SetEstimatedBitrate(250, NowMs());
@@ -181,7 +181,7 @@ TEST_F(ProbeControllerTest, RequestProbeWhenAlrNotEndedRecently) {
   probe_controller_->SetEstimatedBitrate(500, NowMs());
   EXPECT_EQ(probe_controller_->GetAndResetPendingProbes().size(), 2u);
 
-  probe_controller_->SetAlrStartTimeMs(rtc::nullopt);
+  probe_controller_->SetAlrStartTimeMs(absl::nullopt);
   clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
   probe_controller_->Process(NowMs());
   probe_controller_->SetEstimatedBitrate(250, NowMs());

modules/congestion_controller/goog_cc/trendline_estimator.cc

@@ -14,7 +14,7 @@
 
 #include <algorithm>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"
@@ -22,7 +22,7 @@
 namespace webrtc {
 
 namespace {
-rtc::Optional<double> LinearFitSlope(
+absl::optional<double> LinearFitSlope(
     const std::deque<std::pair<double, double>>& points) {
   RTC_DCHECK(points.size() >= 2);
   // Compute the "center of mass".
@@ -42,7 +42,7 @@ rtc::Optional<double> LinearFitSlope(
     denominator += (point.first - x_avg) * (point.first - x_avg);
   }
   if (denominator == 0)
-    return rtc::nullopt;
+    return absl::nullopt;
   return numerator / denominator;
 }
 

modules/congestion_controller/probe_controller.cc

@@ -285,7 +285,7 @@ void ProbeController::Process() {
     return;
 
   // Probe bandwidth periodically when in ALR state.
-  rtc::Optional<int64_t> alr_start_time =
+  absl::optional<int64_t> alr_start_time =
       pacer_->GetApplicationLimitedRegionStartTime();
   if (alr_start_time && estimated_bitrate_bps_ > 0) {
     int64_t next_probe_time_ms =

modules/congestion_controller/probe_controller.h

@@ -79,7 +79,7 @@ class ProbeController {
   int64_t start_bitrate_bps_ RTC_GUARDED_BY(critsect_);
   int64_t max_bitrate_bps_ RTC_GUARDED_BY(critsect_);
   int64_t last_bwe_drop_probing_time_ms_ RTC_GUARDED_BY(critsect_);
-  rtc::Optional<int64_t> alr_end_time_ms_ RTC_GUARDED_BY(critsect_);
+  absl::optional<int64_t> alr_end_time_ms_ RTC_GUARDED_BY(critsect_);
   bool enable_periodic_alr_probing_ RTC_GUARDED_BY(critsect_);
   int64_t time_of_last_large_drop_ms_ RTC_GUARDED_BY(critsect_);
   int64_t bitrate_before_last_large_drop_bps_ RTC_GUARDED_BY(critsect_);

modules/congestion_controller/probe_controller_unittest.cc

@@ -147,7 +147,7 @@ TEST_F(LegacyProbeControllerTest, RequestProbeWhenAlrEndedRecently) {
   testing::Mock::VerifyAndClearExpectations(&pacer_);
   EXPECT_CALL(pacer_, CreateProbeCluster(0.85 * 500)).Times(1);
   EXPECT_CALL(pacer_, GetApplicationLimitedRegionStartTime())
-      .WillRepeatedly(Return(rtc::nullopt));
+      .WillRepeatedly(Return(absl::nullopt));
   clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
   probe_controller_->Process();
   probe_controller_->SetEstimatedBitrate(250);
@@ -164,7 +164,7 @@ TEST_F(LegacyProbeControllerTest, RequestProbeWhenAlrNotEndedRecently) {
   testing::Mock::VerifyAndClearExpectations(&pacer_);
   EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(0);
   EXPECT_CALL(pacer_, GetApplicationLimitedRegionStartTime())
-      .WillRepeatedly(Return(rtc::nullopt));
+      .WillRepeatedly(Return(absl::nullopt));
   clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
   probe_controller_->Process();
   probe_controller_->SetEstimatedBitrate(250);

modules/congestion_controller/rtp/pacer_controller.h

@@ -41,7 +41,7 @@ class PacerController {
   void SetPacerState(bool paused);
   PacedSender* const pacer_;
 
-  rtc::Optional<PacerConfig> current_pacer_config_;
+  absl::optional<PacerConfig> current_pacer_config_;
   bool pacer_paused_ = false;
   bool network_available_ = true;
 

modules/congestion_controller/rtp/send_side_congestion_controller.cc

@@ -185,7 +185,7 @@ class ControlHandler {
   void OnOutstandingData(DataSize in_flight_data);
   void OnPacerQueueUpdate(TimeDelta expected_queue_time);
 
-  rtc::Optional<TargetTransferRate> last_transfer_rate();
+  absl::optional<TargetTransferRate> last_transfer_rate();
 
  private:
   void OnNetworkInvalidation();
@@ -199,8 +199,8 @@ class ControlHandler {
   NetworkChangedObserver* observer_ = nullptr;
   PacerController* pacer_controller_;
 
-  rtc::Optional<TargetTransferRate> current_target_rate_msg_;
+  absl::optional<TargetTransferRate> current_target_rate_msg_;
-  rtc::Optional<DataSize> congestion_window_;
+  absl::optional<DataSize> congestion_window_;
   DataSize outstanding_data_ = DataSize::Zero();
   bool network_available_ = true;
   int64_t last_reported_target_bitrate_bps_ = 0;
@@ -357,7 +357,7 @@ bool ControlHandler::IsSendQueueFull() const {
   return pacer_expected_queue_ms_ > PacedSender::kMaxQueueLengthMs;
 }
 
-rtc::Optional<TargetTransferRate> ControlHandler::last_transfer_rate() {
+absl::optional<TargetTransferRate> ControlHandler::last_transfer_rate() {
   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
   return current_target_rate_msg_;
 }

modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc

@@ -167,7 +167,7 @@ class SendSideCongestionControllerTest : public ::testing::Test {
   PacketRouter packet_router_;
   std::unique_ptr<NiceMock<MockPacedSender>> pacer_;
   std::unique_ptr<SendSideCongestionControllerForTest> controller_;
-  rtc::Optional<uint32_t> target_bitrate_bps_;
+  absl::optional<uint32_t> target_bitrate_bps_;
   std::unique_ptr<rtc::TaskQueue> task_queue_;
 };
 

modules/congestion_controller/rtp/send_time_history.cc

@@ -58,11 +58,11 @@ bool SendTimeHistory::OnSentPacket(uint16_t sequence_number,
   return true;
 }
 
-rtc::Optional<PacketFeedback> SendTimeHistory::GetPacket(
+absl::optional<PacketFeedback> SendTimeHistory::GetPacket(
     uint16_t sequence_number) const {
   int64_t unwrapped_seq_num =
       seq_num_unwrapper_.UnwrapWithoutUpdate(sequence_number);
-  rtc::Optional<PacketFeedback> optional_feedback;
+  absl::optional<PacketFeedback> optional_feedback;
   auto it = history_.find(unwrapped_seq_num);
   if (it != history_.end())
     optional_feedback.emplace(it->second);

modules/congestion_controller/rtp/send_time_history.h

@@ -34,7 +34,7 @@ class SendTimeHistory {
   bool OnSentPacket(uint16_t sequence_number, int64_t send_time_ms);
 
   // Retrieves packet info identified by |sequence_number|.
-  rtc::Optional<PacketFeedback> GetPacket(uint16_t sequence_number) const;
+  absl::optional<PacketFeedback> GetPacket(uint16_t sequence_number) const;
 
   // Look up PacketFeedback for a sent packet, based on the sequence number, and
   // populate all fields except for arrival_time. The packet parameter must
@@ -54,7 +54,7 @@ class SendTimeHistory {
   const int64_t packet_age_limit_ms_;
   SequenceNumberUnwrapper seq_num_unwrapper_;
   std::map<int64_t, PacketFeedback> history_;
-  rtc::Optional<int64_t> last_ack_seq_num_;
+  absl::optional<int64_t> last_ack_seq_num_;
   std::map<RemoteAndLocalNetworkId, size_t> in_flight_bytes_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SendTimeHistory);

modules/congestion_controller/rtp/transport_feedback_adapter.cc

@@ -83,7 +83,7 @@ void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
   send_time_history_.OnSentPacket(sequence_number, send_time_ms);
 }
 
-rtc::Optional<PacketFeedback> TransportFeedbackAdapter::GetPacket(
+absl::optional<PacketFeedback> TransportFeedbackAdapter::GetPacket(
     uint16_t sequence_number) const {
   rtc::CritScope cs(&lock_);
   return send_time_history_.GetPacket(sequence_number);

modules/congestion_controller/rtp/transport_feedback_adapter.h

@@ -48,7 +48,7 @@ class TransportFeedbackAdapter {
   // to the CongestionController interface.
   void OnTransportFeedback(const rtcp::TransportFeedback& feedback);
   std::vector<PacketFeedback> GetTransportFeedbackVector() const;
-  rtc::Optional<PacketFeedback> GetPacket(uint16_t sequence_number) const;
+  absl::optional<PacketFeedback> GetPacket(uint16_t sequence_number) const;
 
   void SetTransportOverhead(int transport_overhead_bytes_per_packet);
 

modules/congestion_controller/send_side_congestion_controller.cc

@@ -385,7 +385,7 @@ void SendSideCongestionController::LimitOutstandingBytes(
     size_t num_outstanding_bytes) {
   RTC_DCHECK(in_cwnd_experiment_);
   rtc::CritScope lock(&network_state_lock_);
-  rtc::Optional<int64_t> min_rtt_ms =
+  absl::optional<int64_t> min_rtt_ms =
       transport_feedback_adapter_.GetMinFeedbackLoopRtt();
   // No valid RTT. Could be because send-side BWE isn't used, in which case
   // we don't try to limit the outstanding packets.

modules/congestion_controller/send_side_congestion_controller_unittest.cc

@@ -142,7 +142,7 @@ class LegacySendSideCongestionControllerTest : public ::testing::Test {
   std::unique_ptr<NiceMock<MockPacedSender>> pacer_;
   std::unique_ptr<SendSideCongestionController> controller_;
 
-  rtc::Optional<uint32_t> target_bitrate_bps_;
+  absl::optional<uint32_t> target_bitrate_bps_;
 };
 
 TEST_F(LegacySendSideCongestionControllerTest, OnNetworkChanged) {

modules/congestion_controller/transport_feedback_adapter.cc

@@ -193,7 +193,8 @@ TransportFeedbackAdapter::GetTransportFeedbackVector() const {
   return last_packet_feedback_vector_;
 }
 
-rtc::Optional<int64_t> TransportFeedbackAdapter::GetMinFeedbackLoopRtt() const {
+absl::optional<int64_t> TransportFeedbackAdapter::GetMinFeedbackLoopRtt()
+    const {
   rtc::CritScope cs(&lock_);
   return min_feedback_rtt_;
 }

modules/congestion_controller/transport_feedback_adapter.h

@@ -47,7 +47,7 @@ class TransportFeedbackAdapter {
   // to the CongestionController interface.
   void OnTransportFeedback(const rtcp::TransportFeedback& feedback);
   std::vector<PacketFeedback> GetTransportFeedbackVector() const;
-  rtc::Optional<int64_t> GetMinFeedbackLoopRtt() const;
+  absl::optional<int64_t> GetMinFeedbackLoopRtt() const;
 
   void SetTransportOverhead(size_t transport_overhead_bytes_per_packet);
 
@@ -68,7 +68,7 @@ class TransportFeedbackAdapter {
   uint16_t local_net_id_ RTC_GUARDED_BY(&lock_);
   uint16_t remote_net_id_ RTC_GUARDED_BY(&lock_);
   std::deque<int64_t> feedback_rtts_ RTC_GUARDED_BY(&lock_);
-  rtc::Optional<int64_t> min_feedback_rtt_ RTC_GUARDED_BY(&lock_);
+  absl::optional<int64_t> min_feedback_rtt_ RTC_GUARDED_BY(&lock_);
 
   rtc::CriticalSection observers_lock_;
   std::vector<PacketFeedbackObserver*> observers_

modules/pacing/BUILD.gn

@@ -38,7 +38,6 @@ rtc_static_library("pacing") {
     "..:module_api",
     "../../:typedefs",
     "../../:webrtc_common",
-    "../../api:optional",
     "../../logging:rtc_event_bwe",
     "../../logging:rtc_event_log_api",
     "../../logging:rtc_event_pacing",
@@ -52,6 +51,7 @@ rtc_static_library("pacing") {
     "../rtp_rtcp",
     "../rtp_rtcp:rtp_rtcp_format",
     "../utility",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 

modules/pacing/alr_detector.cc

@@ -34,7 +34,7 @@ AlrDetector::AlrDetector(RtcEventLog* event_log)
       alr_budget_(0, true),
       event_log_(event_log) {
   RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
-  rtc::Optional<AlrExperimentSettings> experiment_settings =
+  absl::optional<AlrExperimentSettings> experiment_settings =
       AlrExperimentSettings::CreateFromFieldTrial(
           AlrExperimentSettings::kScreenshareProbingBweExperimentName);
   if (!experiment_settings) {
@@ -79,7 +79,7 @@ void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
   alr_budget_.set_target_rate_kbps(rtc::dchecked_cast<int>(target_rate_kbps));
 }
 
-rtc::Optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
+absl::optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
     const {
   return alr_started_time_ms_;
 }

modules/pacing/alr_detector.h

@@ -11,7 +11,7 @@
 #ifndef MODULES_PACING_ALR_DETECTOR_H_
 #define MODULES_PACING_ALR_DETECTOR_H_
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/pacing/interval_budget.h"
 #include "modules/pacing/paced_sender.h"
@@ -42,7 +42,7 @@ class AlrDetector {
 
   // Returns time in milliseconds when the current application-limited region
   // started or empty result if the sender is currently not application-limited.
-  rtc::Optional<int64_t> GetApplicationLimitedRegionStartTime() const;
+  absl::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
 
   // Sent traffic percentage as a function of network capacity used to determine
   // application-limited region. ALR region start when bandwidth usage drops
@@ -62,7 +62,7 @@ class AlrDetector {
   int alr_stop_budget_level_percent_;
 
   IntervalBudget alr_budget_;
-  rtc::Optional<int64_t> alr_started_time_ms_;
+  absl::optional<int64_t> alr_started_time_ms_;
 
   RtcEventLog* event_log_;
 };

modules/pacing/alr_detector_unittest.cc

@@ -60,8 +60,8 @@ class SimulateOutgoingTrafficIn {
     }
   }
   AlrDetector* const alr_detector_;
-  rtc::Optional<int> interval_ms_;
+  absl::optional<int> interval_ms_;
-  rtc::Optional<int> usage_percentage_;
+  absl::optional<int> usage_percentage_;
 };
 }  // namespace
 
@@ -146,7 +146,7 @@ TEST_F(LegacyAlrDetectorTest, BandwidthEstimateChanges) {
 TEST_F(LegacyAlrDetectorTest, ParseControlFieldTrial) {
   webrtc::test::ScopedFieldTrials field_trial(
       "WebRTC-ProbingScreenshareBwe/Control/");
-  rtc::Optional<AlrExperimentSettings> parsed_params =
+  absl::optional<AlrExperimentSettings> parsed_params =
       AlrExperimentSettings::CreateFromFieldTrial(
           "WebRTC-ProbingScreenshareBwe");
   EXPECT_FALSE(static_cast<bool>(parsed_params));
@@ -155,7 +155,7 @@ TEST_F(LegacyAlrDetectorTest, ParseControlFieldTrial) {
 TEST_F(LegacyAlrDetectorTest, ParseActiveFieldTrial) {
   webrtc::test::ScopedFieldTrials field_trial(
       "WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/");
-  rtc::Optional<AlrExperimentSettings> parsed_params =
+  absl::optional<AlrExperimentSettings> parsed_params =
       AlrExperimentSettings::CreateFromFieldTrial(
           "WebRTC-ProbingScreenshareBwe");
   ASSERT_TRUE(static_cast<bool>(parsed_params));

modules/pacing/mock/mock_paced_sender.h

@@ -36,7 +36,7 @@ class MockPacedSender : public PacedSender {
   MOCK_CONST_METHOD0(QueueInPackets, int());
   MOCK_CONST_METHOD0(ExpectedQueueTimeMs, int64_t());
   MOCK_CONST_METHOD0(GetApplicationLimitedRegionStartTime,
-                     rtc::Optional<int64_t>());
+                     absl::optional<int64_t>());
   MOCK_METHOD0(Process, void());
 };
 

modules/pacing/paced_sender.cc

@@ -211,7 +211,7 @@ int64_t PacedSender::ExpectedQueueTimeMs() const {
                               pacing_bitrate_kbps_);
 }
 
-rtc::Optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
+absl::optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
     const {
   rtc::CritScope cs(&critsect_);
   return alr_detector_->GetApplicationLimitedRegionStartTime();

modules/pacing/paced_sender.h

@@ -13,7 +13,7 @@
 
 #include <memory>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "modules/pacing/pacer.h"
 #include "modules/pacing/packet_queue_interface.h"
 #include "rtc_base/criticalsection.h"
@@ -128,7 +128,7 @@ class PacedSender : public Pacer {
   virtual int64_t ExpectedQueueTimeMs() const;
 
   // Deprecated, alr detection will be moved out of the pacer.
-  virtual rtc::Optional<int64_t> GetApplicationLimitedRegionStartTime() const;
+  virtual absl::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
 
   // Returns the number of milliseconds until the module want a worker thread
   // to call Process.

modules/pacing/round_robin_packet_queue.h

@@ -83,8 +83,8 @@ class RoundRobinPacketQueue : public PacketQueueInterface {
 
   const Clock* const clock_;
   int64_t time_last_updated_;
-  rtc::Optional<Packet> pop_packet_;
+  absl::optional<Packet> pop_packet_;
-  rtc::Optional<Stream*> pop_stream_;
+  absl::optional<Stream*> pop_stream_;
 
   bool paused_ = false;
   size_t size_packets_ = 0;

modules/remote_bitrate_estimator/BUILD.gn

@@ -46,7 +46,6 @@ rtc_static_library("remote_bitrate_estimator") {
   deps = [
     "../..:webrtc_common",
     "../../:typedefs",
-    "../../api:optional",
     "../../modules:module_api",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../rtc_base:checks",
@@ -55,6 +54,7 @@ rtc_static_library("remote_bitrate_estimator") {
     "../../system_wrappers",
     "../../system_wrappers:field_trial_api",
     "../../system_wrappers:metrics_api",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 
@@ -135,7 +135,6 @@ if (rtc_include_tests) {
       "..:module_api",
       "../..:webrtc_common",
       "../../:typedefs",
-      "../../api:optional",
       "../../logging:mocks",
       "../../rtc_base:checks",
       "../../rtc_base:gtest_prod",
@@ -161,6 +160,7 @@ if (rtc_include_tests) {
       "../rtp_rtcp:rtp_rtcp_format",
       "../video_coding:video_coding_utility",
       "//testing/gtest",
+      "//third_party/abseil-cpp/absl/types:optional",
     ]
   }
 

modules/remote_bitrate_estimator/aimd_rate_control.cc

@@ -256,7 +256,7 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t new_bitrate_bps,
           // If bitrate decreases more than a normal back off after overuse, it
           // indicates a real network degradation. We do not let such a decrease
           // to determine the bandwidth estimation period.
-          last_decrease_ = rtc::nullopt;
+          last_decrease_ = absl::nullopt;
         } else {
           last_decrease_ = current_bitrate_bps_ - new_bitrate_bps;
         }

modules/remote_bitrate_estimator/aimd_rate_control.h

@@ -87,7 +87,7 @@ class AimdRateControl {
   int64_t rtt_;
   bool in_experiment_;
   bool smoothing_experiment_;
-  rtc::Optional<int> last_decrease_;
+  absl::optional<int> last_decrease_;
 };
 }  // namespace webrtc
 

modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc

@@ -43,7 +43,7 @@ AimdRateControlStates CreateAimdRateControlStates() {
 
 void UpdateRateControl(const AimdRateControlStates& states,
                        const BandwidthUsage& bandwidth_usage,
-                       rtc::Optional<uint32_t> bitrate,
+                       absl::optional<uint32_t> bitrate,
                        int64_t now_ms) {
   RateControlInput input(bandwidth_usage, bitrate, now_ms);
   states.aimd_rate_control->Update(&input, now_ms);
@@ -261,7 +261,7 @@ TEST(AimdRateControlTest, SendingRateBoundedWhenThroughputNotEstimated) {
   UpdateRateControl(states, BandwidthUsage::kBwNormal, kInitialBitrateBps,
                     states.simulated_clock->TimeInMilliseconds());
   for (int i = 0; i < 100; ++i) {
-    UpdateRateControl(states, BandwidthUsage::kBwNormal, rtc::nullopt,
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
                       states.simulated_clock->TimeInMilliseconds());
     states.simulated_clock->AdvanceTimeMilliseconds(100);
   }

modules/remote_bitrate_estimator/bwe_defines.cc

@@ -30,7 +30,7 @@ int GetMinBitrateBps() {
 
 RateControlInput::RateControlInput(
     BandwidthUsage bw_state,
-    const rtc::Optional<uint32_t>& incoming_bitrate,
+    const absl::optional<uint32_t>& incoming_bitrate,
     double noise_var)
     : bw_state(bw_state),
       incoming_bitrate(incoming_bitrate),

modules/remote_bitrate_estimator/include/bwe_defines.h

@@ -11,7 +11,7 @@
 #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
 #define MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
 #define BWE_MAX(a, b) ((a) > (b) ? (a) : (b))
@@ -48,12 +48,12 @@ enum RateControlRegion { kRcNearMax, kRcAboveMax, kRcMaxUnknown };
 
 struct RateControlInput {
   RateControlInput(BandwidthUsage bw_state,
-                   const rtc::Optional<uint32_t>& incoming_bitrate,
+                   const absl::optional<uint32_t>& incoming_bitrate,
                    double noise_var);
   ~RateControlInput();
 
   BandwidthUsage bw_state;
-  rtc::Optional<uint32_t> incoming_bitrate;
+  absl::optional<uint32_t> incoming_bitrate;
   double noise_var;
 };
 }  // namespace webrtc

modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc

@@ -248,7 +248,7 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
   // here.
 
   // Check if incoming bitrate estimate is valid, and if it needs to be reset.
-  rtc::Optional<uint32_t> incoming_bitrate =
+  absl::optional<uint32_t> incoming_bitrate =
       incoming_bitrate_.Rate(arrival_time_ms);
   if (incoming_bitrate) {
     incoming_bitrate_initialized_ = true;
@@ -322,7 +322,7 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
           now_ms - last_update_ms_ > remote_rate_.GetFeedbackInterval()) {
         update_estimate = true;
       } else if (detector_.State() == BandwidthUsage::kBwOverusing) {
-        rtc::Optional<uint32_t> incoming_rate =
+        absl::optional<uint32_t> incoming_rate =
             incoming_bitrate_.Rate(arrival_time_ms);
         if (incoming_rate &&
             remote_rate_.TimeToReduceFurther(now_ms, *incoming_rate)) {

modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc

@@ -97,7 +97,7 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket(
   estimator->last_packet_time_ms = now_ms;
 
   // Check if incoming bitrate estimate is valid, and if it needs to be reset.
-  rtc::Optional<uint32_t> incoming_bitrate = incoming_bitrate_.Rate(now_ms);
+  absl::optional<uint32_t> incoming_bitrate = incoming_bitrate_.Rate(now_ms);
   if (incoming_bitrate) {
     last_valid_incoming_bitrate_ = *incoming_bitrate;
   } else if (last_valid_incoming_bitrate_ > 0) {
@@ -124,7 +124,7 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket(
                                estimator->estimator.num_of_deltas(), now_ms);
   }
   if (estimator->detector.State() == BandwidthUsage::kBwOverusing) {
-    rtc::Optional<uint32_t> incoming_bitrate_bps =
+    absl::optional<uint32_t> incoming_bitrate_bps =
         incoming_bitrate_.Rate(now_ms);
     if (incoming_bitrate_bps &&
         (prior_state != BandwidthUsage::kBwOverusing ||

modules/remote_bitrate_estimator/test/estimators/bbr.cc

@@ -277,15 +277,15 @@ size_t BbrBweSender::TargetCongestionWindow(float gain) {
   return target_congestion_window;
 }
 
-rtc::Optional<int64_t> BbrBweSender::CalculateBandwidthSample(
+absl::optional<int64_t> BbrBweSender::CalculateBandwidthSample(
     size_t data_sent_bytes,
     int64_t send_time_delta_ms,
     size_t data_acked_bytes,
     int64_t ack_time_delta_ms) {
-  rtc::Optional<int64_t> bandwidth_sample;
+  absl::optional<int64_t> bandwidth_sample;
   if (send_time_delta_ms > 0)
     bandwidth_sample.emplace(data_sent_bytes * 8000 / send_time_delta_ms);
-  rtc::Optional<int64_t> ack_rate;
+  absl::optional<int64_t> ack_rate;
   if (ack_time_delta_ms > 0)
     ack_rate.emplace(data_acked_bytes * 8000 / ack_time_delta_ms);
   // If send rate couldn't be calculated automaticaly set |bandwidth_sample| to
@@ -311,7 +311,7 @@ void BbrBweSender::AddSampleForHighGain() {
                             data_acked_before_high_gain_started_bytes_;
   int64_t ack_time_delta_ms = last_packet_ack_time_during_high_gain_ms_ -
                               first_packet_ack_time_during_high_gain_ms_;
-  rtc::Optional<int64_t> bandwidth_sample = CalculateBandwidthSample(
+  absl::optional<int64_t> bandwidth_sample = CalculateBandwidthSample(
       data_sent_bytes, send_time_delta_ms, data_acked_bytes, ack_time_delta_ms);
   if (bandwidth_sample)
     max_bandwidth_filter_->AddBandwidthSample(*bandwidth_sample, round_count_);
@@ -322,7 +322,7 @@ void BbrBweSender::UpdateBandwidthAndMinRtt(
     int64_t now_ms,
     const std::vector<uint16_t>& feedback_vector,
     int64_t bytes_acked) {
-  rtc::Optional<int64_t> min_rtt_sample_ms;
+  absl::optional<int64_t> min_rtt_sample_ms;
   for (uint16_t f : feedback_vector) {
     PacketStats packet = packet_stats_[f];
     size_t data_sent_bytes =
@@ -333,7 +333,7 @@ void BbrBweSender::UpdateBandwidthAndMinRtt(
                               packet.data_acked_before_last_acked_packet_bytes;
     int64_t ack_time_delta_ms =
         packet.ack_time_ms - packet.last_acked_packet_ack_time_ms;
-    rtc::Optional<int64_t> bandwidth_sample =
+    absl::optional<int64_t> bandwidth_sample =
         CalculateBandwidthSample(data_sent_bytes, send_time_delta_ms,
                                  data_acked_bytes, ack_time_delta_ms);
     if (bandwidth_sample)

modules/remote_bitrate_estimator/test/estimators/bbr.h

@@ -17,7 +17,7 @@
 #include <memory>
 #include <vector>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "modules/remote_bitrate_estimator/test/bwe.h"
 #include "rtc_base/numerics/sequence_number_util.h"
 #include "rtc_base/random.h"
@@ -129,10 +129,10 @@ class BbrBweSender : public BweSender {
 
   // Calculates and returns bandwidth sample as minimum between send rate and
   // ack rate, returns nothing if sample cannot be calculated.
-  rtc::Optional<int64_t> CalculateBandwidthSample(size_t data_sent,
+  absl::optional<int64_t> CalculateBandwidthSample(size_t data_sent,
-                                                  int64_t send_time_delta_ms,
+                                                   int64_t send_time_delta_ms,
-                                                  size_t data_acked,
+                                                   size_t data_acked,
-                                                  int64_t ack_time_delta_ms);
+                                                   int64_t ack_time_delta_ms);
 
   // Calculate and add bandwidth sample only for packets' sent during high gain
   // phase. Motivation of having a seperate bucket for high gain phase is to
@@ -174,7 +174,7 @@ class BbrBweSender : public BweSender {
 
   // First moment of time when data inflight decreased below
   // kMinimumCongestionWindow in PROBE_RTT mode.
-  rtc::Optional<int64_t> minimum_congestion_window_start_time_ms_;
+  absl::optional<int64_t> minimum_congestion_window_start_time_ms_;
 
   // First round when data inflight decreased below kMinimumCongestionWindow in
   // PROBE_RTT mode.
@@ -188,7 +188,7 @@ class BbrBweSender : public BweSender {
 
   // Send time of a packet sent first during high gain phase. Also serves as a
   // flag, holding value means that we are already in high gain.
-  rtc::Optional<int64_t> first_packet_send_time_during_high_gain_ms_;
+  absl::optional<int64_t> first_packet_send_time_during_high_gain_ms_;
 
   // Send time of a packet sent last during high gain phase.
   int64_t last_packet_send_time_during_high_gain_ms_;

modules/remote_bitrate_estimator/test/estimators/congestion_window.cc

@@ -33,7 +33,7 @@ CongestionWindow::~CongestionWindow() {}
 
 int CongestionWindow::GetCongestionWindow(BbrBweSender::Mode mode,
                                           int64_t bandwidth_estimate_bps,
-                                          rtc::Optional<int64_t> min_rtt_ms,
+                                          absl::optional<int64_t> min_rtt_ms,
                                           float gain) {
   return GetTargetCongestionWindow(bandwidth_estimate_bps, min_rtt_ms, gain);
 }
@@ -49,7 +49,7 @@ void CongestionWindow::AckReceived(size_t received_packet_size_bytes) {
 
 int CongestionWindow::GetTargetCongestionWindow(
     int64_t bandwidth_estimate_bps,
-    rtc::Optional<int64_t> min_rtt_ms,
+    absl::optional<int64_t> min_rtt_ms,
     float gain) {
   // If we have no rtt sample yet, return the starting congestion window size.
   if (!min_rtt_ms)

modules/remote_bitrate_estimator/test/estimators/congestion_window.h

@@ -14,7 +14,7 @@
 
 #include "modules/remote_bitrate_estimator/test/estimators/bbr.h"
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 
 namespace webrtc {
 namespace testing {
@@ -25,10 +25,10 @@ class CongestionWindow {
   ~CongestionWindow();
   int GetCongestionWindow(BbrBweSender::Mode mode,
                           int64_t bandwidth_estimate,
-                          rtc::Optional<int64_t> min_rtt,
+                          absl::optional<int64_t> min_rtt,
                           float gain);
   int GetTargetCongestionWindow(int64_t bandwidth_estimate,
-                                rtc::Optional<int64_t> min_rtt,
+                                absl::optional<int64_t> min_rtt,
                                 float gain);
   // Packet sent from sender, meaning it is inflight until we receive it and we
   // should add packet's size to data_inflight.

modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h

@@ -16,7 +16,7 @@
 #include <limits>
 #include <list>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 
 namespace webrtc {
 namespace testing {
@@ -36,7 +36,7 @@ class MinRttFilter {
   MinRttFilter() {}
   ~MinRttFilter() {}
 
-  rtc::Optional<int64_t> min_rtt_ms() { return min_rtt_ms_; }
+  absl::optional<int64_t> min_rtt_ms() { return min_rtt_ms_; }
   void AddRttSample(int64_t rtt_ms, int64_t now_ms) {
     if (!min_rtt_ms_ || rtt_ms <= *min_rtt_ms_ || MinRttExpired(now_ms)) {
       min_rtt_ms_.emplace(rtt_ms);
@@ -63,7 +63,7 @@ class MinRttFilter {
   }
 
  private:
-  rtc::Optional<int64_t> min_rtt_ms_;
+  absl::optional<int64_t> min_rtt_ms_;
   std::list<int64_t> rtt_samples_;
 };
 }  // namespace bwe

modules/video_coding/BUILD.gn

@@ -17,7 +17,6 @@ rtc_static_library("encoded_frame") {
   deps = [
     ":video_codec_interface",
     "../../:webrtc_common",
-    "../../api:optional",
     "../../api/video:video_frame_i420",
     "../../common_video:common_video",
     "../../modules:module_api",
@@ -28,6 +27,7 @@ rtc_static_library("encoded_frame") {
     "../../rtc_base/experiments:alr_experiment",
     "../../system_wrappers:field_trial_api",
     "../../system_wrappers:system_wrappers",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 
   if (!build_with_chromium && is_clang) {
@@ -165,7 +165,6 @@ rtc_static_library("video_coding") {
     "../..:webrtc_common",
     "../../:typedefs",
     "../../api:fec_controller_api",
-    "../../api:optional",
     "../../api/video:encoded_frame",
     "../../api/video:video_frame",
     "../../api/video:video_frame_i420",
@@ -185,6 +184,7 @@ rtc_static_library("video_coding") {
     "../../system_wrappers:metrics_api",
     "../rtp_rtcp:rtp_rtcp_format",
     "../utility:utility",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 
@@ -258,7 +258,6 @@ rtc_source_set("video_coding_utility") {
     "..:module_api",
     "../..:webrtc_common",
     "../../:typedefs",
-    "../../api:optional",
     "../../api/video_codecs:video_codecs_api",
     "../../common_video",
     "../../modules/rtp_rtcp",
@@ -270,6 +269,7 @@ rtc_source_set("video_coding_utility") {
     "../../rtc_base/experiments:quality_scaling_experiment",
     "../../system_wrappers",
     "../rtp_rtcp:rtp_rtcp_format",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 
@@ -412,7 +412,6 @@ rtc_static_library("webrtc_vp8_helpers") {
     "..:module_api",
     "../..:webrtc_common",
     "../../:typedefs",
-    "../../api:optional",
     "../../api/video:video_frame",
     "../../api/video_codecs:video_codecs_api",
     "../../common_video",
@@ -422,6 +421,7 @@ rtc_static_library("webrtc_vp8_helpers") {
     "../../system_wrappers",
     "../../system_wrappers:field_trial_api",
     "../../system_wrappers:metrics_api",
+    "//third_party/abseil-cpp/absl/types:optional",
     "//third_party/libyuv",
   ]
 }
@@ -454,7 +454,6 @@ rtc_static_library("webrtc_vp8") {
     "..:module_api",
     "../..:webrtc_common",
     "../../:typedefs",
-    "../../api:optional",
     "../../api/video:video_frame",
     "../../api/video_codecs:video_codecs_api",
     "../../common_video",
@@ -464,6 +463,7 @@ rtc_static_library("webrtc_vp8") {
     "../../system_wrappers",
     "../../system_wrappers:field_trial_api",
     "../../system_wrappers:metrics_api",
+    "//third_party/abseil-cpp/absl/types:optional",
     "//third_party/libyuv",
   ]
   if (rtc_build_libvpx) {
@@ -758,7 +758,6 @@ if (rtc_include_tests) {
       "../..:webrtc_common",
       "../../api:create_videocodec_test_fixture_api",
       "../../api:mock_video_codec_factory",
-      "../../api:optional",
       "../../api:videocodec_test_fixture_api",
       "../../api/video:video_frame_i420",
       "../../api/video_codecs:rtc_software_fallback_wrappers",
@@ -773,6 +772,7 @@ if (rtc_include_tests) {
       "../../test:test_support",
       "../../test:video_test_common",
       "../rtp_rtcp:rtp_rtcp_format",
+      "//third_party/abseil-cpp/absl/types:optional",
     ]
 
     data = video_coding_modules_tests_resources

modules/video_coding/codecs/h264/h264.cc

@@ -43,7 +43,7 @@ bool IsH264CodecSupported() {
 SdpVideoFormat CreateH264Format(H264::Profile profile,
                                 H264::Level level,
                                 const std::string& packetization_mode) {
-  const rtc::Optional<std::string> profile_string =
+  const absl::optional<std::string> profile_string =
       H264::ProfileLevelIdToString(H264::ProfileLevelId(profile, level));
   RTC_CHECK(profile_string);
   return SdpVideoFormat(

modules/video_coding/codecs/h264/h264_decoder_impl.cc

@@ -325,7 +325,7 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
   RTC_CHECK_EQ(av_frame_->data[kVPlaneIndex], i420_buffer->DataV());
   video_frame->set_timestamp(input_image._timeStamp);
 
-  rtc::Optional<uint8_t> qp;
+  absl::optional<uint8_t> qp;
   // TODO(sakal): Maybe it is possible to get QP directly from FFmpeg.
   h264_bitstream_parser_.ParseBitstream(input_image._buffer,
                                         input_image._length);
@@ -351,10 +351,10 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
         video_frame->rotation());
     // TODO(nisse): Timestamp and rotation are all zero here. Change decoder
     // interface to pass a VideoFrameBuffer instead of a VideoFrame?
-    decoded_image_callback_->Decoded(cropped_frame, rtc::nullopt, qp);
+    decoded_image_callback_->Decoded(cropped_frame, absl::nullopt, qp);
   } else {
     // Return decoded frame.
-    decoded_image_callback_->Decoded(*video_frame, rtc::nullopt, qp);
+    decoded_image_callback_->Decoded(*video_frame, absl::nullopt, qp);
   }
   // Stop referencing it, possibly freeing |video_frame|.
   av_frame_unref(av_frame_.get());

modules/video_coding/codecs/h264/test/h264_impl_unittest.cc

@@ -50,7 +50,7 @@ TEST_F(TestH264Impl, MAYBE_EncodeDecode) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
             decoder_->Decode(encoded_frame, false, nullptr, 0));
   std::unique_ptr<VideoFrame> decoded_frame;
-  rtc::Optional<uint8_t> decoded_qp;
+  absl::optional<uint8_t> decoded_qp;
   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
   ASSERT_TRUE(decoded_frame);
   EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
@@ -67,7 +67,7 @@ TEST_F(TestH264Impl, MAYBE_DecodedQpEqualsEncodedQp) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
             decoder_->Decode(encoded_frame, false, nullptr, 0));
   std::unique_ptr<VideoFrame> decoded_frame;
-  rtc::Optional<uint8_t> decoded_qp;
+  absl::optional<uint8_t> decoded_qp;
   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
   ASSERT_TRUE(decoded_frame);
   ASSERT_TRUE(decoded_qp);

modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h

@@ -42,8 +42,8 @@ class MultiplexDecoderAdapter : public VideoDecoder {
 
   void Decoded(AlphaCodecStream stream_idx,
                VideoFrame* decoded_image,
-               rtc::Optional<int32_t> decode_time_ms,
+               absl::optional<int32_t> decode_time_ms,
-               rtc::Optional<uint8_t> qp);
+               absl::optional<uint8_t> qp);
 
  private:
   // Wrapper class that redirects Decoded() calls.
@@ -53,11 +53,11 @@ class MultiplexDecoderAdapter : public VideoDecoder {
   struct DecodedImageData;
 
   void MergeAlphaImages(VideoFrame* decoded_image,
-                        const rtc::Optional<int32_t>& decode_time_ms,
+                        const absl::optional<int32_t>& decode_time_ms,
-                        const rtc::Optional<uint8_t>& qp,
+                        const absl::optional<uint8_t>& qp,
                         VideoFrame* multiplex_decoded_image,
-                        const rtc::Optional<int32_t>& multiplex_decode_time_ms,
+                        const absl::optional<int32_t>& multiplex_decode_time_ms,
-                        const rtc::Optional<uint8_t>& multiplex_qp);
+                        const absl::optional<uint8_t>& multiplex_qp);
 
   VideoDecoderFactory* const factory_;
   const SdpVideoFormat associated_format_;

modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc

@@ -33,8 +33,8 @@ class MultiplexDecoderAdapter::AdapterDecodedImageCallback
       : adapter_(adapter), stream_idx_(stream_idx) {}
 
   void Decoded(VideoFrame& decoded_image,
-               rtc::Optional<int32_t> decode_time_ms,
+               absl::optional<int32_t> decode_time_ms,
-               rtc::Optional<uint8_t> qp) override {
+               absl::optional<uint8_t> qp) override {
     if (!adapter_)
       return;
     adapter_->Decoded(stream_idx_, &decoded_image, decode_time_ms, qp);
@@ -64,16 +64,16 @@ struct MultiplexDecoderAdapter::DecodedImageData {
   }
   DecodedImageData(AlphaCodecStream stream_idx,
                    const VideoFrame& decoded_image,
-                   const rtc::Optional<int32_t>& decode_time_ms,
+                   const absl::optional<int32_t>& decode_time_ms,
-                   const rtc::Optional<uint8_t>& qp)
+                   const absl::optional<uint8_t>& qp)
       : stream_idx_(stream_idx),
         decoded_image_(decoded_image),
         decode_time_ms_(decode_time_ms),
         qp_(qp) {}
   const AlphaCodecStream stream_idx_;
   VideoFrame decoded_image_;
-  const rtc::Optional<int32_t> decode_time_ms_;
+  const absl::optional<int32_t> decode_time_ms_;
-  const rtc::Optional<uint8_t> qp_;
+  const absl::optional<uint8_t> qp_;
 
  private:
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DecodedImageData);
@@ -153,8 +153,8 @@ int32_t MultiplexDecoderAdapter::Release() {
 
 void MultiplexDecoderAdapter::Decoded(AlphaCodecStream stream_idx,
                                       VideoFrame* decoded_image,
-                                      rtc::Optional<int32_t> decode_time_ms,
+                                      absl::optional<int32_t> decode_time_ms,
-                                      rtc::Optional<uint8_t> qp) {
+                                      absl::optional<uint8_t> qp) {
   const auto& other_decoded_data_it =
       decoded_data_.find(decoded_image->timestamp());
   if (other_decoded_data_it != decoded_data_.end()) {
@@ -184,11 +184,11 @@ void MultiplexDecoderAdapter::Decoded(AlphaCodecStream stream_idx,
 
 void MultiplexDecoderAdapter::MergeAlphaImages(
     VideoFrame* decoded_image,
-    const rtc::Optional<int32_t>& decode_time_ms,
+    const absl::optional<int32_t>& decode_time_ms,
-    const rtc::Optional<uint8_t>& qp,
+    const absl::optional<uint8_t>& qp,
     VideoFrame* alpha_decoded_image,
-    const rtc::Optional<int32_t>& alpha_decode_time_ms,
+    const absl::optional<int32_t>& alpha_decode_time_ms,
-    const rtc::Optional<uint8_t>& alpha_qp) {
+    const absl::optional<uint8_t>& alpha_qp) {
   if (!alpha_decoded_image->timestamp()) {
     decoded_complete_callback_->Decoded(*decoded_image, decode_time_ms, qp);
     return;

modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc

@@ -132,7 +132,7 @@ TEST_F(TestMultiplexAdapter, EncodeDecodeI420Frame) {
       WEBRTC_VIDEO_CODEC_OK,
       decoder_->Decode(encoded_frame, false, &codec_specific_info, -1));
   std::unique_ptr<VideoFrame> decoded_frame;
-  rtc::Optional<uint8_t> decoded_qp;
+  absl::optional<uint8_t> decoded_qp;
   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
   ASSERT_TRUE(decoded_frame);
   EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
@@ -150,7 +150,7 @@ TEST_F(TestMultiplexAdapter, EncodeDecodeI420AFrame) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
             decoder_->Decode(encoded_frame, false, nullptr, 0));
   std::unique_ptr<VideoFrame> decoded_frame;
-  rtc::Optional<uint8_t> decoded_qp;
+  absl::optional<uint8_t> decoded_qp;
   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
   ASSERT_TRUE(decoded_frame);
   EXPECT_GT(I420PSNR(yuva_frame.get(), decoded_frame.get()), 36);

modules/video_coding/codecs/test/video_codec_unittest.cc

@@ -50,8 +50,8 @@ VideoCodecUnitTest::FakeEncodeCompleteCallback::OnEncodedImage(
 
 void VideoCodecUnitTest::FakeDecodeCompleteCallback::Decoded(
     VideoFrame& frame,
-    rtc::Optional<int32_t> decode_time_ms,
+    absl::optional<int32_t> decode_time_ms,
-    rtc::Optional<uint8_t> qp) {
+    absl::optional<uint8_t> qp) {
   rtc::CritScope lock(&test_->decoded_frame_section_);
   test_->decoded_frame_.emplace(frame);
   test_->decoded_qp_ = qp;
@@ -71,7 +71,8 @@ void VideoCodecUnitTest::SetUp() {
 
   input_frame_generator_ = test::FrameGenerator::CreateSquareGenerator(
       codec_settings_.width, codec_settings_.height,
-      rtc::Optional<test::FrameGenerator::OutputType>(), rtc::Optional<int>());
+      absl::optional<test::FrameGenerator::OutputType>(),
+      absl::optional<int>());
 
   encoder_ = CreateEncoder();
   decoder_ = CreateDecoder();
@@ -141,7 +142,7 @@ bool VideoCodecUnitTest::WaitForEncodedFrames(
 }
 
 bool VideoCodecUnitTest::WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
-                                             rtc::Optional<uint8_t>* qp) {
+                                             absl::optional<uint8_t>* qp) {
   bool ret = decoded_frame_event_.Wait(kDecodeTimeoutMs);
   EXPECT_TRUE(ret) << "Timed out while waiting for a decoded frame.";
   // This becomes unsafe if there are multiple threads waiting for frames.

modules/video_coding/codecs/test/video_codec_unittest.h

@@ -67,8 +67,8 @@ class VideoCodecUnitTest : public ::testing::Test {
       return -1;
     }
     void Decoded(VideoFrame& frame,
-                 rtc::Optional<int32_t> decode_time_ms,
+                 absl::optional<int32_t> decode_time_ms,
-                 rtc::Optional<uint8_t> qp) override;
+                 absl::optional<uint8_t> qp) override;
 
    private:
     VideoCodecUnitTest* const test_;
@@ -97,7 +97,7 @@ class VideoCodecUnitTest : public ::testing::Test {
 
   // Helper method for waiting a single decoded frame.
   bool WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
-                           rtc::Optional<uint8_t>* qp);
+                           absl::optional<uint8_t>* qp);
 
   size_t GetNumEncodedFrames();
 
@@ -120,9 +120,9 @@ class VideoCodecUnitTest : public ::testing::Test {
 
   rtc::Event decoded_frame_event_;
   rtc::CriticalSection decoded_frame_section_;
-  rtc::Optional<VideoFrame> decoded_frame_
+  absl::optional<VideoFrame> decoded_frame_
       RTC_GUARDED_BY(decoded_frame_section_);
-  rtc::Optional<uint8_t> decoded_qp_ RTC_GUARDED_BY(decoded_frame_section_);
+  absl::optional<uint8_t> decoded_qp_ RTC_GUARDED_BY(decoded_frame_section_);
 
   std::unique_ptr<test::FrameGenerator> input_frame_generator_;
   uint32_t last_input_frame_timestamp_;

modules/video_coding/codecs/test/videoprocessor.h

@@ -151,8 +151,8 @@ class VideoProcessor {
     }
 
     void Decoded(webrtc::VideoFrame& image,
-                 rtc::Optional<int32_t> decode_time_ms,
+                 absl::optional<int32_t> decode_time_ms,
-                 rtc::Optional<uint8_t> qp) override {
+                 absl::optional<uint8_t> qp) override {
       Decoded(image);
     }
 

modules/video_coding/codecs/vp8/default_temporal_layers.h

@@ -17,7 +17,7 @@
 
 #include "modules/video_coding/codecs/vp8/temporal_layers.h"
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 
 namespace webrtc {
 
@@ -52,7 +52,7 @@ class DefaultTemporalLayers : public TemporalLayers {
   uint8_t pattern_idx_;
   bool last_base_layer_sync_;
   // Updated cumulative bitrates, per temporal layer.
-  rtc::Optional<std::vector<uint32_t>> new_bitrates_bps_;
+  absl::optional<std::vector<uint32_t>> new_bitrates_bps_;
 };
 
 class DefaultTemporalLayersChecker : public TemporalLayersChecker {

modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc

@@ -308,7 +308,7 @@ int LibvpxVp8Decoder::ReturnFrame(const vpx_image_t* img,
 
   VideoFrame decoded_image(buffer, timestamp, 0, kVideoRotation_0);
   decoded_image.set_ntp_time_ms(ntp_time_ms);
-  decode_complete_callback_->Decoded(decoded_image, rtc::nullopt, qp);
+  decode_complete_callback_->Decoded(decoded_image, absl::nullopt, qp);
 
   return WEBRTC_VIDEO_CODEC_OK;
 }

modules/video_coding/codecs/vp8/screenshare_layers.h

@@ -72,9 +72,9 @@ class ScreenshareLayers : public TemporalLayers {
   uint32_t max_debt_bytes_;
 
   // Configured max framerate.
-  rtc::Optional<uint32_t> target_framerate_;
+  absl::optional<uint32_t> target_framerate_;
   // Incoming framerate from capturer.
-  rtc::Optional<uint32_t> capture_framerate_;
+  absl::optional<uint32_t> capture_framerate_;
   // Tracks what framerate we actually encode, and drops frames on overshoot.
   RateStatistics encode_framerate_;
   bool bitrate_updated_;

modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc

@@ -138,11 +138,11 @@ class ScreenshareLayerTest : public ::testing::Test {
   // FrameEncoded() call will be omitted and needs to be done by the caller.
   // Returns the flags for the last frame.
   int SkipUntilTl(int layer) {
-    return SkipUntilTlAndSync(layer, rtc::nullopt);
+    return SkipUntilTlAndSync(layer, absl::nullopt);
   }
 
   // Same as SkipUntilTl, but also waits until the sync bit condition is met.
-  int SkipUntilTlAndSync(int layer, rtc::Optional<bool> sync) {
+  int SkipUntilTlAndSync(int layer, absl::optional<bool> sync) {
     int flags = 0;
     const int kMaxFramesToSkip =
         1 + (sync.value_or(false) ? kMaxSyncPeriodSeconds : 1) * kFrameRate;

modules/video_coding/codecs/vp8/simulcast_test_fixture_impl.cc

@@ -150,8 +150,8 @@ class SimulcastTestFixtureImpl::Vp8TestDecodedImageCallback
     return -1;
   }
   void Decoded(VideoFrame& decoded_image,
-               rtc::Optional<int32_t> decode_time_ms,
+               absl::optional<int32_t> decode_time_ms,
-               rtc::Optional<uint8_t> qp) override {
+               absl::optional<uint8_t> qp) override {
     Decoded(decoded_image);
   }
   int DecodedFrames() { return decoded_frames_; }

modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc

@@ -162,7 +162,7 @@ TEST_F(TestVp8Impl, DecodedQpEqualsEncodedQp) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
             decoder_->Decode(encoded_frame, false, nullptr, -1));
   std::unique_ptr<VideoFrame> decoded_frame;
-  rtc::Optional<uint8_t> decoded_qp;
+  absl::optional<uint8_t> decoded_qp;
   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
   ASSERT_TRUE(decoded_frame);
   ASSERT_TRUE(decoded_qp);
@@ -249,7 +249,7 @@ TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
             decoder_->Decode(encoded_frame, false, nullptr, -1));
 
   std::unique_ptr<VideoFrame> decoded_frame;
-  rtc::Optional<uint8_t> decoded_qp;
+  absl::optional<uint8_t> decoded_qp;
   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
   ASSERT_TRUE(decoded_frame);
   // Compute PSNR on all planes (faster than SSIM).
@@ -283,7 +283,7 @@ TEST_F(TestVp8Impl, MAYBE_DecodeWithACompleteKeyFrame) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
             decoder_->Decode(encoded_frame, false, nullptr, -1));
   std::unique_ptr<VideoFrame> decoded_frame;
-  rtc::Optional<uint8_t> decoded_qp;
+  absl::optional<uint8_t> decoded_qp;
   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
   ASSERT_TRUE(decoded_frame);
   EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);

modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc

@@ -80,7 +80,7 @@ TEST_F(TestVp9Impl, EncodeDecode) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
             decoder_->Decode(encoded_frame, false, nullptr, 0));
   std::unique_ptr<VideoFrame> decoded_frame;
-  rtc::Optional<uint8_t> decoded_qp;
+  absl::optional<uint8_t> decoded_qp;
   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
   ASSERT_TRUE(decoded_frame);
   EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
@@ -118,7 +118,7 @@ TEST_F(TestVp9Impl, DecodedQpEqualsEncodedQp) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
             decoder_->Decode(encoded_frame, false, nullptr, 0));
   std::unique_ptr<VideoFrame> decoded_frame;
-  rtc::Optional<uint8_t> decoded_qp;
+  absl::optional<uint8_t> decoded_qp;
   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
   ASSERT_TRUE(decoded_frame);
   ASSERT_TRUE(decoded_qp);

modules/video_coding/codecs/vp9/vp9_impl.cc

@@ -1101,7 +1101,7 @@ int VP9DecoderImpl::ReturnFrame(const vpx_image_t* img,
                            0 /* render_time_ms */, webrtc::kVideoRotation_0);
   decoded_image.set_ntp_time_ms(ntp_time_ms);
 
-  decode_complete_callback_->Decoded(decoded_image, rtc::nullopt, qp);
+  decode_complete_callback_->Decoded(decoded_image, absl::nullopt, qp);
   return WEBRTC_VIDEO_CODEC_OK;
 }
 

modules/video_coding/codecs/vp9/vp9_impl.h

@@ -113,7 +113,7 @@ class VP9EncoderImpl : public VP9Encoder {
   InterLayerPredMode inter_layer_pred_;
 
   // Framerate controller.
-  rtc::Optional<float> target_framerate_fps_;
+  absl::optional<float> target_framerate_fps_;
   RateStatistics output_framerate_;
   uint32_t last_encoded_frame_rtp_timestamp_;
 

modules/video_coding/frame_buffer2.cc

@@ -599,7 +599,7 @@ void FrameBuffer::UpdateJitterDelay() {
 
 void FrameBuffer::UpdateTimingFrameInfo() {
   TRACE_EVENT0("webrtc", "FrameBuffer::UpdateTimingFrameInfo");
-  rtc::Optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo();
+  absl::optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo();
   if (info && stats_callback_)
     stats_callback_->OnTimingFrameInfoUpdated(*info);
 }

modules/video_coding/frame_object.cc

@@ -156,11 +156,11 @@ bool RtpFrameObject::delayed_by_retransmission() const {
   return times_nacked() > 0;
 }
 
-rtc::Optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
+absl::optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
   rtc::CritScope lock(&packet_buffer_->crit_);
   VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
   if (!packet)
-    return rtc::nullopt;
+    return absl::nullopt;
   return packet->video_header.codecHeader;
 }
 

modules/video_coding/frame_object.h

@@ -11,7 +11,7 @@
 #ifndef MODULES_VIDEO_CODING_FRAME_OBJECT_H_
 #define MODULES_VIDEO_CODING_FRAME_OBJECT_H_
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "api/video/encoded_frame.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/include/module_common_types.h"
@@ -41,7 +41,7 @@ class RtpFrameObject : public EncodedFrame {
   int64_t ReceivedTime() const override;
   int64_t RenderTime() const override;
   bool delayed_by_retransmission() const override;
-  rtc::Optional<RTPVideoTypeHeader> GetCodecHeader() const;
+  absl::optional<RTPVideoTypeHeader> GetCodecHeader() const;
 
  private:
   rtc::scoped_refptr<PacketBuffer> packet_buffer_;

modules/video_coding/generic_decoder.cc

@@ -57,15 +57,15 @@ int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage) {
 int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
                                          int64_t decode_time_ms) {
   Decoded(decodedImage,
-          decode_time_ms >= 0 ? rtc::Optional<int32_t>(decode_time_ms)
+          decode_time_ms >= 0 ? absl::optional<int32_t>(decode_time_ms)
-                              : rtc::nullopt,
+                              : absl::nullopt,
-          rtc::nullopt);
+          absl::nullopt);
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
 void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
-                                      rtc::Optional<int32_t> decode_time_ms,
+                                      absl::optional<int32_t> decode_time_ms,
-                                      rtc::Optional<uint8_t> qp) {
+                                      absl::optional<uint8_t> qp) {
   RTC_DCHECK(_receiveCallback) << "Callback must not be null at this point";
   TRACE_EVENT_INSTANT1("webrtc", "VCMDecodedFrameCallback::Decoded",
                        "timestamp", decodedImage.timestamp());

modules/video_coding/generic_decoder.h

@@ -46,8 +46,8 @@ class VCMDecodedFrameCallback : public DecodedImageCallback {
   int32_t Decoded(VideoFrame& decodedImage) override;
   int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) override;
   void Decoded(VideoFrame& decodedImage,
-               rtc::Optional<int32_t> decode_time_ms,
+               absl::optional<int32_t> decode_time_ms,
-               rtc::Optional<uint8_t> qp) override;
+               absl::optional<uint8_t> qp) override;
   int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId) override;
   int32_t ReceivedDecodedFrame(const uint64_t pictureId) override;
 

modules/video_coding/generic_encoder.cc

@@ -12,7 +12,7 @@
 
 #include <vector>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "api/video/i420_buffer.h"
 #include "modules/include/module_common_types_public.h"
 #include "modules/video_coding/encoded_frame.h"
@@ -178,7 +178,7 @@ VCMEncodedFrameCallback::VCMEncodedFrameCallback(
       incorrect_capture_time_logged_messages_(0),
       reordered_frames_logged_messages_(0),
       stalled_encoder_logged_messages_(0) {
-  rtc::Optional<AlrExperimentSettings> experiment_settings =
+  absl::optional<AlrExperimentSettings> experiment_settings =
       AlrExperimentSettings::CreateFromFieldTrial(
           AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
   if (experiment_settings) {
@@ -249,10 +249,10 @@ void VCMEncodedFrameCallback::OnEncodeStarted(uint32_t rtp_timestamp,
       rtp_timestamp, capture_time_ms, rtc::TimeMillis());
 }
 
-rtc::Optional<int64_t> VCMEncodedFrameCallback::ExtractEncodeStartTime(
+absl::optional<int64_t> VCMEncodedFrameCallback::ExtractEncodeStartTime(
     size_t simulcast_svc_idx,
     EncodedImage* encoded_image) {
-  rtc::Optional<int64_t> result;
+  absl::optional<int64_t> result;
   size_t num_simulcast_svc_streams = timing_frames_info_.size();
   if (simulcast_svc_idx < num_simulcast_svc_streams) {
     auto encode_start_list =
@@ -308,8 +308,8 @@ rtc::Optional<int64_t> VCMEncodedFrameCallback::ExtractEncodeStartTime(
 
 void VCMEncodedFrameCallback::FillTimingInfo(size_t simulcast_svc_idx,
                                              EncodedImage* encoded_image) {
-  rtc::Optional<size_t> outlier_frame_size;
+  absl::optional<size_t> outlier_frame_size;
-  rtc::Optional<int64_t> encode_start_ms;
+  absl::optional<int64_t> encode_start_ms;
   uint8_t timing_flags = VideoSendTiming::kNotTriggered;
   {
     rtc::CritScope crit(&timing_params_lock_);

modules/video_coding/generic_encoder.h

@@ -79,8 +79,8 @@ class VCMEncodedFrameCallback : public EncodedImageCallback {
  private:
   // For non-internal-source encoders, returns encode started time and fixes
   // capture timestamp for the frame, if corrupted by the encoder.
-  rtc::Optional<int64_t> ExtractEncodeStartTime(size_t simulcast_svc_idx,
+  absl::optional<int64_t> ExtractEncodeStartTime(size_t simulcast_svc_idx,
-                                                EncodedImage* encoded_image)
+                                                 EncodedImage* encoded_image)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(timing_params_lock_);
 
   void FillTimingInfo(size_t simulcast_svc_idx, EncodedImage* encoded_image);

modules/video_coding/h264_sps_pps_tracker.cc

@@ -219,9 +219,9 @@ void H264SpsPpsTracker::InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
     RTC_LOG(LS_WARNING) << "SPS Nalu header missing";
     return;
   }
-  rtc::Optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
+  absl::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
       sps.data() + kNaluHeaderOffset, sps.size() - kNaluHeaderOffset);
-  rtc::Optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
+  absl::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
       pps.data() + kNaluHeaderOffset, pps.size() - kNaluHeaderOffset);
 
   if (!parsed_sps) {

modules/video_coding/include/mock/mock_vcm_callbacks.h

@@ -34,7 +34,7 @@ class MockVCMReceiveCallback : public VCMReceiveCallback {
   virtual ~MockVCMReceiveCallback() {}
 
   MOCK_METHOD3(FrameToRender,
-               int32_t(VideoFrame&, rtc::Optional<uint8_t>, VideoContentType));
+               int32_t(VideoFrame&, absl::optional<uint8_t>, VideoContentType));
   MOCK_METHOD1(ReceivedDecodedReferenceFrame, int32_t(const uint64_t));
   MOCK_METHOD1(OnIncomingPayloadType, void(int));
   MOCK_METHOD1(OnDecoderImplementationName, void(const char*));

modules/video_coding/include/mock/mock_video_codec_interface.h

@@ -58,8 +58,8 @@ class MockDecodedImageCallback : public DecodedImageCallback {
                        int64_t decode_time_ms));
   MOCK_METHOD3(Decoded,
                void(VideoFrame& decodedImage,  // NOLINT
-                    rtc::Optional<int32_t> decode_time_ms,
+                    absl::optional<int32_t> decode_time_ms,
-                    rtc::Optional<uint8_t> qp));
+                    absl::optional<uint8_t> qp));
   MOCK_METHOD1(ReceivedDecodedReferenceFrame,
                int32_t(const uint64_t pictureId));
   MOCK_METHOD1(ReceivedDecodedFrame, int32_t(const uint64_t pictureId));

modules/video_coding/include/video_coding_defines.h

@@ -70,7 +70,7 @@ struct VCMFrameCount {
 class VCMReceiveCallback {
  public:
   virtual int32_t FrameToRender(VideoFrame& videoFrame,  // NOLINT
-                                rtc::Optional<uint8_t> qp,
+                                absl::optional<uint8_t> qp,
                                 VideoContentType content_type) = 0;
 
   virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId);

modules/video_coding/packet_buffer.cc

@@ -199,12 +199,12 @@ void PacketBuffer::PaddingReceived(uint16_t seq_num) {
     received_frame_callback_->OnReceivedFrame(std::move(frame));
 }
 
-rtc::Optional<int64_t> PacketBuffer::LastReceivedPacketMs() const {
+absl::optional<int64_t> PacketBuffer::LastReceivedPacketMs() const {
   rtc::CritScope lock(&crit_);
   return last_received_packet_ms_;
 }
 
-rtc::Optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
+absl::optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
   rtc::CritScope lock(&crit_);
   return last_received_keyframe_packet_ms_;
 }

modules/video_coding/packet_buffer.h

@@ -58,8 +58,8 @@ class PacketBuffer {
   void PaddingReceived(uint16_t seq_num);
 
   // Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
-  rtc::Optional<int64_t> LastReceivedPacketMs() const;
+  absl::optional<int64_t> LastReceivedPacketMs() const;
-  rtc::Optional<int64_t> LastReceivedKeyframePacketMs() const;
+  absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
 
   // Returns number of different frames seen in the packet buffer
   int GetUniqueFramesSeen() const;
@@ -159,13 +159,13 @@ class PacketBuffer {
   OnReceivedFrameCallback* const received_frame_callback_;
 
   // Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
-  rtc::Optional<int64_t> last_received_packet_ms_ RTC_GUARDED_BY(crit_);
+  absl::optional<int64_t> last_received_packet_ms_ RTC_GUARDED_BY(crit_);
-  rtc::Optional<int64_t> last_received_keyframe_packet_ms_
+  absl::optional<int64_t> last_received_keyframe_packet_ms_
       RTC_GUARDED_BY(crit_);
 
   int unique_frames_seen_ RTC_GUARDED_BY(crit_);
 
-  rtc::Optional<uint16_t> newest_inserted_seq_num_ RTC_GUARDED_BY(crit_);
+  absl::optional<uint16_t> newest_inserted_seq_num_ RTC_GUARDED_BY(crit_);
   std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_
       RTC_GUARDED_BY(crit_);
 

modules/video_coding/rtp_frame_reference_finder.cc

@@ -240,7 +240,7 @@ RtpFrameReferenceFinder::ManageFrameGeneric(RtpFrameObject* frame,
 
 RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
     RtpFrameObject* frame) {
-  rtc::Optional<RTPVideoTypeHeader> rtp_codec_header = frame->GetCodecHeader();
+  absl::optional<RTPVideoTypeHeader> rtp_codec_header = frame->GetCodecHeader();
   if (!rtp_codec_header) {
     RTC_LOG(LS_WARNING)
         << "Failed to get codec header from frame, dropping frame.";
@@ -393,7 +393,7 @@ void RtpFrameReferenceFinder::UpdateLayerInfoVp8(
 
 RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp9(
     RtpFrameObject* frame) {
-  rtc::Optional<RTPVideoTypeHeader> rtp_codec_header = frame->GetCodecHeader();
+  absl::optional<RTPVideoTypeHeader> rtp_codec_header = frame->GetCodecHeader();
   if (!rtp_codec_header) {
     RTC_LOG(LS_WARNING)
         << "Failed to get codec header from frame, dropping frame.";

modules/video_coding/timing.cc

@@ -239,7 +239,7 @@ void VCMTiming::SetTimingFrameInfo(const TimingFrameInfo& info) {
   timing_frame_info_.emplace(info);
 }
 
-rtc::Optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
+absl::optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
   rtc::CritScope cs(&crit_sect_);
   return timing_frame_info_;
 }

modules/video_coding/timing.h

@@ -95,7 +95,7 @@ class VCMTiming {
                           int* render_delay_ms) const;
 
   void SetTimingFrameInfo(const TimingFrameInfo& info);
-  rtc::Optional<TimingFrameInfo> GetTimingFrameInfo();
+  absl::optional<TimingFrameInfo> GetTimingFrameInfo();
 
   enum { kDefaultRenderDelayMs = 10 };
   enum { kDelayMaxChangeMsPerS = 100 };
@@ -124,7 +124,7 @@ class VCMTiming {
   int current_delay_ms_ RTC_GUARDED_BY(crit_sect_);
   int last_decode_ms_ RTC_GUARDED_BY(crit_sect_);
   uint32_t prev_frame_timestamp_ RTC_GUARDED_BY(crit_sect_);
-  rtc::Optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(crit_sect_);
+  absl::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(crit_sect_);
   size_t num_decoded_frames_ RTC_GUARDED_BY(crit_sect_);
 };
 }  // namespace webrtc

modules/video_coding/utility/moving_average.cc

@@ -23,13 +23,13 @@ void MovingAverage::AddSample(int sample) {
   sum_history_[count_ % sum_history_.size()] = sum_;
 }
 
-rtc::Optional<int> MovingAverage::GetAverage() const {
+absl::optional<int> MovingAverage::GetAverage() const {
   return GetAverage(size());
 }
 
-rtc::Optional<int> MovingAverage::GetAverage(size_t num_samples) const {
+absl::optional<int> MovingAverage::GetAverage(size_t num_samples) const {
   if (num_samples > size() || num_samples == 0)
-    return rtc::nullopt;
+    return absl::nullopt;
   int sum = sum_ - sum_history_[(count_ - num_samples) % sum_history_.size()];
   return sum / static_cast<int>(num_samples);
 }

modules/video_coding/utility/moving_average.h

@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 
 namespace webrtc {
 class MovingAverage {
@@ -21,8 +21,8 @@ class MovingAverage {
   explicit MovingAverage(size_t s);
   ~MovingAverage();
   void AddSample(int sample);
-  rtc::Optional<int> GetAverage() const;
+  absl::optional<int> GetAverage() const;
-  rtc::Optional<int> GetAverage(size_t num_samples) const;
+  absl::optional<int> GetAverage(size_t num_samples) const;
   void Reset();
   size_t size() const;