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}diff --git a/modules/congestion_controller/bbr/BUILD.gn b/modules/congestion_controller/bbr/BUILD.gn
index 4ddf2c9..73d6eb1 100644
--- a/modules/congestion_controller/bbr/BUILD.gn
+++ b/modules/congestion_controller/bbr/BUILD.gn
@@ -31,12 +31,12 @@
":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 @@
]
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 @@
"loss_rate_filter.h",
]
deps = [
- "../../../api:optional",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("rtt_stats") {
diff --git a/modules/congestion_controller/bbr/bandwidth_sampler.h b/modules/congestion_controller/bbr/bandwidth_sampler.h
index 9f448a6..7e0a44e 100644
--- a/modules/congestion_controller/bbr/bandwidth_sampler.h
+++ b/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 @@
// 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 @@
// 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_;
diff --git a/modules/congestion_controller/bbr/bbr_network_controller.cc b/modules/congestion_controller/bbr/bbr_network_controller.cc
index 4bbcfac..a8de48a 100644
--- a/modules/congestion_controller/bbr/bbr_network_controller.cc
+++ b/modules/congestion_controller/bbr/bbr_network_controller.cc
@@ -382,7 +382,7 @@
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";
diff --git a/modules/congestion_controller/bbr/bbr_network_controller.h b/modules/congestion_controller/bbr/bbr_network_controller.h
index 20d12c8..788a0d4 100644
--- a/modules/congestion_controller/bbr/bbr_network_controller.h
+++ b/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 @@
webrtc::Random random_;
LossRateFilter loss_rate_;
- rtc::Optional<TargetRateConstraints> constraints_;
+ absl::optional<TargetRateConstraints> constraints_;
Mode mode_;
@@ -244,7 +244,7 @@
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 @@
// 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 @@
// 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_;
diff --git a/modules/congestion_controller/bbr/loss_rate_filter.h b/modules/congestion_controller/bbr/loss_rate_filter.h
index 4cc0cbe..6e31b63 100644
--- a/modules/congestion_controller/bbr/loss_rate_filter.h
+++ b/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 {
diff --git a/modules/congestion_controller/delay_based_bwe.cc b/modules/congestion_controller/delay_based_bwe.cc
index a28cc7e..ceca36d 100644
--- a/modules/congestion_controller/delay_based_bwe.cc
+++ b/modules/congestion_controller/delay_based_bwe.cc
@@ -114,7 +114,7 @@
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 @@
}
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 @@
}
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.
diff --git a/modules/congestion_controller/delay_based_bwe.h b/modules/congestion_controller/delay_based_bwe.h
index 315af01..3326490 100644
--- a/modules/congestion_controller/delay_based_bwe.h
+++ b/modules/congestion_controller/delay_based_bwe.h
@@ -47,7 +47,7 @@
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 @@
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_;
diff --git a/modules/congestion_controller/delay_based_bwe_unittest.cc b/modules/congestion_controller/delay_based_bwe_unittest.cc
index f7b0c74..88dc68c 100644
--- a/modules/congestion_controller/delay_based_bwe_unittest.cc
+++ b/modules/congestion_controller/delay_based_bwe_unittest.cc
@@ -29,7 +29,7 @@
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 @@
PacketFeedback::kNoSendTime,
1, 1500, PacedPacketInfo()));
bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector,
- rtc::nullopt);
+ absl::nullopt);
}
TEST_F(LegacyDelayBasedBweTest, ProbeDetection) {
diff --git a/modules/congestion_controller/goog_cc/BUILD.gn b/modules/congestion_controller/goog_cc/BUILD.gn
index 789770a..fad65ef 100644
--- a/modules/congestion_controller/goog_cc/BUILD.gn
+++ b/modules/congestion_controller/goog_cc/BUILD.gn
@@ -43,7 +43,6 @@
"../..:module_api",
"../../..:webrtc_common",
"../../../:typedefs",
- "../../../api:optional",
"../../../api/transport:network_control",
"../../../logging:rtc_event_log_api",
"../../../logging:rtc_event_pacing",
@@ -57,6 +56,7 @@
"../../pacing",
"../../remote_bitrate_estimator",
"../../rtp_rtcp:rtp_rtcp_format",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -85,7 +85,6 @@
}
deps = [
- "../../../api:optional",
"../../../logging:rtc_event_bwe",
"../../../logging:rtc_event_log_api",
"../../../rtc_base:checks",
@@ -96,6 +95,7 @@
"../../../system_wrappers:metrics_api",
"../../remote_bitrate_estimator",
"../../rtp_rtcp:rtp_rtcp_format",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
index 49d1b3a..c5d4289 100644
--- a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
+++ b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
@@ -47,7 +47,7 @@
}
}
-rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
+absl::optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
return bitrate_estimator_->bitrate_bps();
}
diff --git a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
index c545a4a..faa6b48 100644
--- a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
+++ b/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 @@
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_;
};
diff --git a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
index 38f8df7..92a24c6 100644
--- a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
+++ b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
@@ -35,7 +35,7 @@
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, 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));
diff --git a/modules/congestion_controller/goog_cc/alr_detector.cc b/modules/congestion_controller/goog_cc/alr_detector.cc
index f36b1f9..941809d 100644
--- a/modules/congestion_controller/goog_cc/alr_detector.cc
+++ b/modules/congestion_controller/goog_cc/alr_detector.cc
@@ -35,7 +35,7 @@
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 @@
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_;
}
diff --git a/modules/congestion_controller/goog_cc/alr_detector.h b/modules/congestion_controller/goog_cc/alr_detector.h
index 98293a7..36fb952 100644
--- a/modules/congestion_controller/goog_cc/alr_detector.h
+++ b/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 @@
// 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 @@
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_;
};
diff --git a/modules/congestion_controller/goog_cc/alr_detector_unittest.cc b/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
index 9aeab51..0aaa2e1 100644
--- a/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
+++ b/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
@@ -65,8 +65,8 @@
}
AlrDetector* const alr_detector_;
int64_t* timestamp_ms_;
- rtc::Optional<int> interval_ms_;
- rtc::Optional<int> usage_percentage_;
+ absl::optional<int> interval_ms_;
+ absl::optional<int> usage_percentage_;
};
} // namespace
@@ -152,7 +152,7 @@
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, 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));
diff --git a/modules/congestion_controller/goog_cc/bitrate_estimator.cc b/modules/congestion_controller/goog_cc/bitrate_estimator.cc
index 5280b38..692c3da 100644
--- a/modules/congestion_controller/goog_cc/bitrate_estimator.cc
+++ b/modules/congestion_controller/goog_cc/bitrate_estimator.cc
@@ -92,9 +92,9 @@
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;
}
diff --git a/modules/congestion_controller/goog_cc/bitrate_estimator.h b/modules/congestion_controller/goog_cc/bitrate_estimator.h
index f49ace4..24f08fa 100644
--- a/modules/congestion_controller/goog_cc/bitrate_estimator.h
+++ b/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 @@
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();
diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe.cc b/modules/congestion_controller/goog_cc/delay_based_bwe.cc
index fea5a66..818af94 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe.cc
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe.cc
@@ -114,7 +114,7 @@
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 @@
}
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 @@
}
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.
diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe.h b/modules/congestion_controller/goog_cc/delay_based_bwe.h
index 485f687..08f59c8 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe.h
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe.h
@@ -48,7 +48,7 @@
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 @@
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_;
diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
index b19e8f9..a4afcaf 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
@@ -31,7 +31,7 @@
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 @@
PacketFeedback::kNoSendTime,
1, 1500, PacedPacketInfo()));
bitrate_estimator_->IncomingPacketFeedbackVector(
- packet_feedback_vector, rtc::nullopt, kDummyTimestamp);
+ packet_feedback_vector, absl::nullopt, kDummyTimestamp);
}
TEST_F(DelayBasedBweTest, ProbeDetection) {
diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
index 829faea..8c00ad8 100644
--- a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
+++ b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
@@ -101,7 +101,7 @@
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::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::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 @@
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 @@
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 =
diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control.h b/modules/congestion_controller/goog_cc/goog_cc_network_control.h
index 3afd8d2..ba66fce 100644
--- a/modules/congestion_controller/goog_cc/goog_cc_network_control.h
+++ b/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 @@
private:
void UpdateBitrateConstraints(TargetRateConstraints constraints,
- rtc::Optional<DataRate> starting_rate);
- rtc::Optional<DataSize> MaybeUpdateCongestionWindow();
+ absl::optional<DataRate> starting_rate);
+ absl::optional<DataSize> MaybeUpdateCongestionWindow();
NetworkControlUpdate MaybeTriggerOnNetworkChanged(Timestamp at_time);
bool GetNetworkParameters(int32_t* estimated_bitrate_bps,
uint8_t* fraction_loss,
@@ -71,10 +71,10 @@
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 @@
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);
};
diff --git a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
index c929306..17e4aca 100644
--- a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
+++ b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
@@ -170,9 +170,9 @@
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;
}
diff --git a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
index 55738d8..ed92521 100644
--- a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
+++ b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
@@ -28,7 +28,7 @@
// 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 @@
std::map<int, AggregatedCluster> clusters_;
RtcEventLog* const event_log_;
- rtc::Optional<int> estimated_bitrate_bps_;
+ absl::optional<int> estimated_bitrate_bps_;
};
} // namespace webrtc
diff --git a/modules/congestion_controller/goog_cc/probe_controller.cc b/modules/congestion_controller/goog_cc/probe_controller.cc
index c05c094..fbc0620 100644
--- a/modules/congestion_controller/goog_cc/probe_controller.cc
+++ b/modules/congestion_controller/goog_cc/probe_controller.cc
@@ -209,7 +209,7 @@
}
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) {
diff --git a/modules/congestion_controller/goog_cc/probe_controller.h b/modules/congestion_controller/goog_cc/probe_controller.h
index 163ccb2c..c8946e8 100644
--- a/modules/congestion_controller/goog_cc/probe_controller.h
+++ b/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 @@
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 @@
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_;
- rtc::Optional<int64_t> alr_end_time_ms_;
+ absl::optional<int64_t> alr_start_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_;
diff --git a/modules/congestion_controller/goog_cc/probe_controller_unittest.cc b/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
index 4a61e67..ffa131d 100644
--- a/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
+++ b/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
@@ -161,7 +161,7 @@
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 @@
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());
diff --git a/modules/congestion_controller/goog_cc/trendline_estimator.cc b/modules/congestion_controller/goog_cc/trendline_estimator.cc
index d69ea3f..14aae95 100644
--- a/modules/congestion_controller/goog_cc/trendline_estimator.cc
+++ b/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 @@
denominator += (point.first - x_avg) * (point.first - x_avg);
}
if (denominator == 0)
- return rtc::nullopt;
+ return absl::nullopt;
return numerator / denominator;
}
diff --git a/modules/congestion_controller/probe_controller.cc b/modules/congestion_controller/probe_controller.cc
index ea1f45c..56fc8ad 100644
--- a/modules/congestion_controller/probe_controller.cc
+++ b/modules/congestion_controller/probe_controller.cc
@@ -285,7 +285,7 @@
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 =
diff --git a/modules/congestion_controller/probe_controller.h b/modules/congestion_controller/probe_controller.h
index 084055b..5bc16bc 100644
--- a/modules/congestion_controller/probe_controller.h
+++ b/modules/congestion_controller/probe_controller.h
@@ -79,7 +79,7 @@
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_);
diff --git a/modules/congestion_controller/probe_controller_unittest.cc b/modules/congestion_controller/probe_controller_unittest.cc
index 796599a..6e47e01 100644
--- a/modules/congestion_controller/probe_controller_unittest.cc
+++ b/modules/congestion_controller/probe_controller_unittest.cc
@@ -147,7 +147,7 @@
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 @@
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);
diff --git a/modules/congestion_controller/rtp/pacer_controller.h b/modules/congestion_controller/rtp/pacer_controller.h
index 27760c4..1b87e36 100644
--- a/modules/congestion_controller/rtp/pacer_controller.h
+++ b/modules/congestion_controller/rtp/pacer_controller.h
@@ -41,7 +41,7 @@
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;
diff --git a/modules/congestion_controller/rtp/send_side_congestion_controller.cc b/modules/congestion_controller/rtp/send_side_congestion_controller.cc
index ace522c..6e0af24 100644
--- a/modules/congestion_controller/rtp/send_side_congestion_controller.cc
+++ b/modules/congestion_controller/rtp/send_side_congestion_controller.cc
@@ -185,7 +185,7 @@
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 @@
NetworkChangedObserver* observer_ = nullptr;
PacerController* pacer_controller_;
- rtc::Optional<TargetTransferRate> current_target_rate_msg_;
- rtc::Optional<DataSize> congestion_window_;
+ absl::optional<TargetTransferRate> current_target_rate_msg_;
+ 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 @@
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_;
}
diff --git a/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc b/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc
index 136a989..df0c62a 100644
--- a/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc
+++ b/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc
@@ -167,7 +167,7 @@
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_;
};
diff --git a/modules/congestion_controller/rtp/send_time_history.cc b/modules/congestion_controller/rtp/send_time_history.cc
index c83d371..73682cd 100644
--- a/modules/congestion_controller/rtp/send_time_history.cc
+++ b/modules/congestion_controller/rtp/send_time_history.cc
@@ -58,11 +58,11 @@
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);
diff --git a/modules/congestion_controller/rtp/send_time_history.h b/modules/congestion_controller/rtp/send_time_history.h
index f2d4b9f..7cc6ad7 100644
--- a/modules/congestion_controller/rtp/send_time_history.h
+++ b/modules/congestion_controller/rtp/send_time_history.h
@@ -34,7 +34,7 @@
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 @@
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);
diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/modules/congestion_controller/rtp/transport_feedback_adapter.cc
index d7c7bf9..d1b4855 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter.cc
@@ -83,7 +83,7 @@
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);
diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter.h b/modules/congestion_controller/rtp/transport_feedback_adapter.h
index 67c00f5..ae38b84 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter.h
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter.h
@@ -48,7 +48,7 @@
// 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);
diff --git a/modules/congestion_controller/send_side_congestion_controller.cc b/modules/congestion_controller/send_side_congestion_controller.cc
index 0fd4ca5..8df5ed0 100644
--- a/modules/congestion_controller/send_side_congestion_controller.cc
+++ b/modules/congestion_controller/send_side_congestion_controller.cc
@@ -385,7 +385,7 @@
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.
diff --git a/modules/congestion_controller/send_side_congestion_controller_unittest.cc b/modules/congestion_controller/send_side_congestion_controller_unittest.cc
index 4c91fe4..3653b8c 100644
--- a/modules/congestion_controller/send_side_congestion_controller_unittest.cc
+++ b/modules/congestion_controller/send_side_congestion_controller_unittest.cc
@@ -142,7 +142,7 @@
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) {
diff --git a/modules/congestion_controller/transport_feedback_adapter.cc b/modules/congestion_controller/transport_feedback_adapter.cc
index 7a5f7c4..3baa3ad 100644
--- a/modules/congestion_controller/transport_feedback_adapter.cc
+++ b/modules/congestion_controller/transport_feedback_adapter.cc
@@ -193,7 +193,8 @@
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_;
}
diff --git a/modules/congestion_controller/transport_feedback_adapter.h b/modules/congestion_controller/transport_feedback_adapter.h
index faab4d0..b411c26 100644
--- a/modules/congestion_controller/transport_feedback_adapter.h
+++ b/modules/congestion_controller/transport_feedback_adapter.h
@@ -47,7 +47,7 @@
// 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 @@
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_
diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn
index 4ac56d2..a8b6e3f 100644
--- a/modules/pacing/BUILD.gn
+++ b/modules/pacing/BUILD.gn
@@ -38,7 +38,6 @@
"..:module_api",
"../../:typedefs",
"../../:webrtc_common",
- "../../api:optional",
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../logging:rtc_event_pacing",
@@ -52,6 +51,7 @@
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
"../utility",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/pacing/alr_detector.cc b/modules/pacing/alr_detector.cc
index 9ba6e32..17fae29 100644
--- a/modules/pacing/alr_detector.cc
+++ b/modules/pacing/alr_detector.cc
@@ -34,7 +34,7 @@
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 @@
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_;
}
diff --git a/modules/pacing/alr_detector.h b/modules/pacing/alr_detector.h
index f73bc17..cffd6c0 100644
--- a/modules/pacing/alr_detector.h
+++ b/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 @@
// 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 @@
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_;
};
diff --git a/modules/pacing/alr_detector_unittest.cc b/modules/pacing/alr_detector_unittest.cc
index f892fbd..13eb65f 100644
--- a/modules/pacing/alr_detector_unittest.cc
+++ b/modules/pacing/alr_detector_unittest.cc
@@ -60,8 +60,8 @@
}
}
AlrDetector* const alr_detector_;
- rtc::Optional<int> interval_ms_;
- rtc::Optional<int> usage_percentage_;
+ absl::optional<int> interval_ms_;
+ absl::optional<int> usage_percentage_;
};
} // namespace
@@ -146,7 +146,7 @@
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, 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));
diff --git a/modules/pacing/mock/mock_paced_sender.h b/modules/pacing/mock/mock_paced_sender.h
index b153862..da4fd04 100644
--- a/modules/pacing/mock/mock_paced_sender.h
+++ b/modules/pacing/mock/mock_paced_sender.h
@@ -36,7 +36,7 @@
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());
};
diff --git a/modules/pacing/paced_sender.cc b/modules/pacing/paced_sender.cc
index 00440e4..276e4d3 100644
--- a/modules/pacing/paced_sender.cc
+++ b/modules/pacing/paced_sender.cc
@@ -211,7 +211,7 @@
pacing_bitrate_kbps_);
}
-rtc::Optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
+absl::optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
const {
rtc::CritScope cs(&critsect_);
return alr_detector_->GetApplicationLimitedRegionStartTime();
diff --git a/modules/pacing/paced_sender.h b/modules/pacing/paced_sender.h
index 8f02e78..8b40792 100644
--- a/modules/pacing/paced_sender.h
+++ b/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 @@
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.
diff --git a/modules/pacing/round_robin_packet_queue.h b/modules/pacing/round_robin_packet_queue.h
index fe0cf70..8351b80 100644
--- a/modules/pacing/round_robin_packet_queue.h
+++ b/modules/pacing/round_robin_packet_queue.h
@@ -83,8 +83,8 @@
const Clock* const clock_;
int64_t time_last_updated_;
- rtc::Optional<Packet> pop_packet_;
- rtc::Optional<Stream*> pop_stream_;
+ absl::optional<Packet> pop_packet_;
+ absl::optional<Stream*> pop_stream_;
bool paused_ = false;
size_t size_packets_ = 0;
diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn
index 31eb46c..b157f3d 100644
--- a/modules/remote_bitrate_estimator/BUILD.gn
+++ b/modules/remote_bitrate_estimator/BUILD.gn
@@ -46,7 +46,6 @@
deps = [
"../..:webrtc_common",
"../../:typedefs",
- "../../api:optional",
"../../modules:module_api",
"../../modules/rtp_rtcp:rtp_rtcp_format",
"../../rtc_base:checks",
@@ -55,6 +54,7 @@
"../../system_wrappers",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -135,7 +135,6 @@
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
- "../../api:optional",
"../../logging:mocks",
"../../rtc_base:checks",
"../../rtc_base:gtest_prod",
@@ -161,6 +160,7 @@
"../rtp_rtcp:rtp_rtcp_format",
"../video_coding:video_coding_utility",
"//testing/gtest",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control.cc b/modules/remote_bitrate_estimator/aimd_rate_control.cc
index 52a41e4..949cc6a 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control.cc
+++ b/modules/remote_bitrate_estimator/aimd_rate_control.cc
@@ -256,7 +256,7 @@
// 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;
}
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control.h b/modules/remote_bitrate_estimator/aimd_rate_control.h
index 9791c1d4..c62ad7d 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control.h
+++ b/modules/remote_bitrate_estimator/aimd_rate_control.h
@@ -87,7 +87,7 @@
int64_t rtt_;
bool in_experiment_;
bool smoothing_experiment_;
- rtc::Optional<int> last_decrease_;
+ absl::optional<int> last_decrease_;
};
} // namespace webrtc
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc b/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
index de88682..a561cdf 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
+++ b/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
@@ -43,7 +43,7 @@
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 @@
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);
}
diff --git a/modules/remote_bitrate_estimator/bwe_defines.cc b/modules/remote_bitrate_estimator/bwe_defines.cc
index b4e6885..af2ec76 100644
--- a/modules/remote_bitrate_estimator/bwe_defines.cc
+++ b/modules/remote_bitrate_estimator/bwe_defines.cc
@@ -30,7 +30,7 @@
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),
diff --git a/modules/remote_bitrate_estimator/include/bwe_defines.h b/modules/remote_bitrate_estimator/include/bwe_defines.h
index 9b5b48b..de393b6 100644
--- a/modules/remote_bitrate_estimator/include/bwe_defines.h
+++ b/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 @@
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
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index fdc1523..b061cbd 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -248,7 +248,7 @@
// 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 @@
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)) {
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
index a914a84..612a51e 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
@@ -97,7 +97,7 @@
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 @@
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 ||
diff --git a/modules/remote_bitrate_estimator/test/estimators/bbr.cc b/modules/remote_bitrate_estimator/test/estimators/bbr.cc
index c9ed3ae..d6209e0 100644
--- a/modules/remote_bitrate_estimator/test/estimators/bbr.cc
+++ b/modules/remote_bitrate_estimator/test/estimators/bbr.cc
@@ -277,15 +277,15 @@
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 @@
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 @@
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 @@
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)
diff --git a/modules/remote_bitrate_estimator/test/estimators/bbr.h b/modules/remote_bitrate_estimator/test/estimators/bbr.h
index 966f1e2..1d16ac5 100644
--- a/modules/remote_bitrate_estimator/test/estimators/bbr.h
+++ b/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 @@
// 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,
- int64_t send_time_delta_ms,
- size_t data_acked,
- int64_t ack_time_delta_ms);
+ absl::optional<int64_t> CalculateBandwidthSample(size_t data_sent,
+ int64_t send_time_delta_ms,
+ size_t data_acked,
+ 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 @@
// 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 @@
// 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_;
diff --git a/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc b/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc
index f3f66a0..ac22cd0 100644
--- a/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc
+++ b/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc
@@ -33,7 +33,7 @@
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 @@
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)
diff --git a/modules/remote_bitrate_estimator/test/estimators/congestion_window.h b/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
index f344adb..a0952af 100644
--- a/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
+++ b/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 @@
~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.
diff --git a/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h b/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
index ed8ec86..f7e1e14 100644
--- a/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
+++ b/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 @@
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 @@
}
private:
- rtc::Optional<int64_t> min_rtt_ms_;
+ absl::optional<int64_t> min_rtt_ms_;
std::list<int64_t> rtt_samples_;
};
} // namespace bwe
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 18baa68..90ab60a 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -17,7 +17,6 @@
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_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 @@
"../..: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 @@
"../../system_wrappers:metrics_api",
"../rtp_rtcp:rtp_rtcp_format",
"../utility:utility",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -258,7 +258,6 @@
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
- "../../api:optional",
"../../api/video_codecs:video_codecs_api",
"../../common_video",
"../../modules/rtp_rtcp",
@@ -270,6 +269,7 @@
"../../rtc_base/experiments:quality_scaling_experiment",
"../../system_wrappers",
"../rtp_rtcp:rtp_rtcp_format",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -412,7 +412,6 @@
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
- "../../api:optional",
"../../api/video:video_frame",
"../../api/video_codecs:video_codecs_api",
"../../common_video",
@@ -422,6 +421,7 @@
"../../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 @@
"..:module_api",
"../..:webrtc_common",
"../../:typedefs",
- "../../api:optional",
"../../api/video:video_frame",
"../../api/video_codecs:video_codecs_api",
"../../common_video",
@@ -464,6 +463,7 @@
"../../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 @@
"../..: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 @@
"../../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
diff --git a/modules/video_coding/codecs/h264/h264.cc b/modules/video_coding/codecs/h264/h264.cc
index 4408fab..93d4750 100644
--- a/modules/video_coding/codecs/h264/h264.cc
+++ b/modules/video_coding/codecs/h264/h264.cc
@@ -43,7 +43,7 @@
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(
diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
index c560056..a2be91c 100644
--- a/modules/video_coding/codecs/h264/h264_decoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
@@ -325,7 +325,7 @@
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 @@
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());
diff --git a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
index a9d790e..88ef5f2 100644
--- a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
+++ b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
@@ -50,7 +50,7 @@
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 @@
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);
diff --git a/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h b/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h
index 6694e11..09618a2 100644
--- a/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h
+++ b/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h
@@ -42,8 +42,8 @@
void Decoded(AlphaCodecStream stream_idx,
VideoFrame* decoded_image,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp);
+ absl::optional<int32_t> decode_time_ms,
+ absl::optional<uint8_t> qp);
private:
// Wrapper class that redirects Decoded() calls.
@@ -53,11 +53,11 @@
struct DecodedImageData;
void MergeAlphaImages(VideoFrame* decoded_image,
- const rtc::Optional<int32_t>& decode_time_ms,
- const rtc::Optional<uint8_t>& qp,
+ const absl::optional<int32_t>& decode_time_ms,
+ const absl::optional<uint8_t>& qp,
VideoFrame* multiplex_decoded_image,
- const rtc::Optional<int32_t>& multiplex_decode_time_ms,
- const rtc::Optional<uint8_t>& multiplex_qp);
+ const absl::optional<int32_t>& multiplex_decode_time_ms,
+ const absl::optional<uint8_t>& multiplex_qp);
VideoDecoderFactory* const factory_;
const SdpVideoFormat associated_format_;
diff --git a/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc b/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
index 2a485b7..de4352f 100644
--- a/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
+++ b/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
@@ -33,8 +33,8 @@
: adapter_(adapter), stream_idx_(stream_idx) {}
void Decoded(VideoFrame& decoded_image,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp) override {
+ absl::optional<int32_t> decode_time_ms,
+ absl::optional<uint8_t> qp) override {
if (!adapter_)
return;
adapter_->Decoded(stream_idx_, &decoded_image, decode_time_ms, qp);
@@ -64,16 +64,16 @@
}
DecodedImageData(AlphaCodecStream stream_idx,
const VideoFrame& decoded_image,
- const rtc::Optional<int32_t>& decode_time_ms,
- const rtc::Optional<uint8_t>& qp)
+ const absl::optional<int32_t>& decode_time_ms,
+ 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 rtc::Optional<uint8_t> qp_;
+ const absl::optional<int32_t> decode_time_ms_;
+ const absl::optional<uint8_t> qp_;
private:
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DecodedImageData);
@@ -153,8 +153,8 @@
void MultiplexDecoderAdapter::Decoded(AlphaCodecStream stream_idx,
VideoFrame* decoded_image,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp) {
+ absl::optional<int32_t> decode_time_ms,
+ 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::MergeAlphaImages(
VideoFrame* decoded_image,
- const rtc::Optional<int32_t>& decode_time_ms,
- const rtc::Optional<uint8_t>& qp,
+ const absl::optional<int32_t>& decode_time_ms,
+ const absl::optional<uint8_t>& qp,
VideoFrame* alpha_decoded_image,
- const rtc::Optional<int32_t>& alpha_decode_time_ms,
- const rtc::Optional<uint8_t>& alpha_qp) {
+ const absl::optional<int32_t>& alpha_decode_time_ms,
+ const absl::optional<uint8_t>& alpha_qp) {
if (!alpha_decoded_image->timestamp()) {
decoded_complete_callback_->Decoded(*decoded_image, decode_time_ms, qp);
return;
diff --git a/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc b/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc
index 49afe01..fbefd60 100644
--- a/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc
+++ b/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc
@@ -132,7 +132,7 @@
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 @@
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);
diff --git a/modules/video_coding/codecs/test/video_codec_unittest.cc b/modules/video_coding/codecs/test/video_codec_unittest.cc
index bd65fa6..5027163 100644
--- a/modules/video_coding/codecs/test/video_codec_unittest.cc
+++ b/modules/video_coding/codecs/test/video_codec_unittest.cc
@@ -50,8 +50,8 @@
void VideoCodecUnitTest::FakeDecodeCompleteCallback::Decoded(
VideoFrame& frame,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp) {
+ absl::optional<int32_t> decode_time_ms,
+ 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 @@
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::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.
diff --git a/modules/video_coding/codecs/test/video_codec_unittest.h b/modules/video_coding/codecs/test/video_codec_unittest.h
index b5917e3..c880bc8 100644
--- a/modules/video_coding/codecs/test/video_codec_unittest.h
+++ b/modules/video_coding/codecs/test/video_codec_unittest.h
@@ -67,8 +67,8 @@
return -1;
}
void Decoded(VideoFrame& frame,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp) override;
+ absl::optional<int32_t> decode_time_ms,
+ absl::optional<uint8_t> qp) override;
private:
VideoCodecUnitTest* const test_;
@@ -97,7 +97,7 @@
// 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 @@
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_;
diff --git a/modules/video_coding/codecs/test/videoprocessor.h b/modules/video_coding/codecs/test/videoprocessor.h
index ae90e11..d44ee1f 100644
--- a/modules/video_coding/codecs/test/videoprocessor.h
+++ b/modules/video_coding/codecs/test/videoprocessor.h
@@ -151,8 +151,8 @@
}
void Decoded(webrtc::VideoFrame& image,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp) override {
+ absl::optional<int32_t> decode_time_ms,
+ absl::optional<uint8_t> qp) override {
Decoded(image);
}
diff --git a/modules/video_coding/codecs/vp8/default_temporal_layers.h b/modules/video_coding/codecs/vp8/default_temporal_layers.h
index f8bb09b..2c3c87f 100644
--- a/modules/video_coding/codecs/vp8/default_temporal_layers.h
+++ b/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 @@
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 {
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
index 0524eae..bd2c992 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
@@ -308,7 +308,7 @@
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;
}
diff --git a/modules/video_coding/codecs/vp8/screenshare_layers.h b/modules/video_coding/codecs/vp8/screenshare_layers.h
index c479917..c1b5fa7 100644
--- a/modules/video_coding/codecs/vp8/screenshare_layers.h
+++ b/modules/video_coding/codecs/vp8/screenshare_layers.h
@@ -72,9 +72,9 @@
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_;
diff --git a/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
index e6f121b..82d176a 100644
--- a/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
+++ b/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
@@ -138,11 +138,11 @@
// 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;
diff --git a/modules/video_coding/codecs/vp8/simulcast_test_fixture_impl.cc b/modules/video_coding/codecs/vp8/simulcast_test_fixture_impl.cc
index 361a5a6..5a4712f 100644
--- a/modules/video_coding/codecs/vp8/simulcast_test_fixture_impl.cc
+++ b/modules/video_coding/codecs/vp8/simulcast_test_fixture_impl.cc
@@ -150,8 +150,8 @@
return -1;
}
void Decoded(VideoFrame& decoded_image,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp) override {
+ absl::optional<int32_t> decode_time_ms,
+ absl::optional<uint8_t> qp) override {
Decoded(decoded_image);
}
int DecodedFrames() { return decoded_frames_; }
diff --git a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index 207a078..a3d7cd9 100644
--- a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -162,7 +162,7 @@
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 @@
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 @@
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);
diff --git a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
index 09bd36d..5eb2447 100644
--- a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
+++ b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
@@ -80,7 +80,7 @@
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 @@
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);
diff --git a/modules/video_coding/codecs/vp9/vp9_impl.cc b/modules/video_coding/codecs/vp9/vp9_impl.cc
index 29c47cb..7a1e2ab 100644
--- a/modules/video_coding/codecs/vp9/vp9_impl.cc
+++ b/modules/video_coding/codecs/vp9/vp9_impl.cc
@@ -1101,7 +1101,7 @@
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;
}
diff --git a/modules/video_coding/codecs/vp9/vp9_impl.h b/modules/video_coding/codecs/vp9/vp9_impl.h
index 76a3e0b..53ed671 100644
--- a/modules/video_coding/codecs/vp9/vp9_impl.h
+++ b/modules/video_coding/codecs/vp9/vp9_impl.h
@@ -113,7 +113,7 @@
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_;
diff --git a/modules/video_coding/frame_buffer2.cc b/modules/video_coding/frame_buffer2.cc
index 37feae1..f13ed35 100644
--- a/modules/video_coding/frame_buffer2.cc
+++ b/modules/video_coding/frame_buffer2.cc
@@ -599,7 +599,7 @@
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);
}
diff --git a/modules/video_coding/frame_object.cc b/modules/video_coding/frame_object.cc
index 676175d..a3fc0d1 100644
--- a/modules/video_coding/frame_object.cc
+++ b/modules/video_coding/frame_object.cc
@@ -156,11 +156,11 @@
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;
}
diff --git a/modules/video_coding/frame_object.h b/modules/video_coding/frame_object.h
index 69c1cde..8980984 100644
--- a/modules/video_coding/frame_object.h
+++ b/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 @@
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_;
diff --git a/modules/video_coding/generic_decoder.cc b/modules/video_coding/generic_decoder.cc
index 1d1c2fd..580bfc9 100644
--- a/modules/video_coding/generic_decoder.cc
+++ b/modules/video_coding/generic_decoder.cc
@@ -57,15 +57,15 @@
int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
int64_t decode_time_ms) {
Decoded(decodedImage,
- decode_time_ms >= 0 ? rtc::Optional<int32_t>(decode_time_ms)
- : rtc::nullopt,
- rtc::nullopt);
+ decode_time_ms >= 0 ? absl::optional<int32_t>(decode_time_ms)
+ : absl::nullopt,
+ absl::nullopt);
return WEBRTC_VIDEO_CODEC_OK;
}
void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp) {
+ absl::optional<int32_t> decode_time_ms,
+ 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());
diff --git a/modules/video_coding/generic_decoder.h b/modules/video_coding/generic_decoder.h
index 2d940d8..8137b20 100644
--- a/modules/video_coding/generic_decoder.h
+++ b/modules/video_coding/generic_decoder.h
@@ -46,8 +46,8 @@
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,
- rtc::Optional<uint8_t> qp) override;
+ absl::optional<int32_t> decode_time_ms,
+ absl::optional<uint8_t> qp) override;
int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId) override;
int32_t ReceivedDecodedFrame(const uint64_t pictureId) override;
diff --git a/modules/video_coding/generic_encoder.cc b/modules/video_coding/generic_encoder.cc
index 69357ee..202cc86 100644
--- a/modules/video_coding/generic_encoder.cc
+++ b/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 @@
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 @@
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 @@
void VCMEncodedFrameCallback::FillTimingInfo(size_t simulcast_svc_idx,
EncodedImage* encoded_image) {
- rtc::Optional<size_t> outlier_frame_size;
- rtc::Optional<int64_t> encode_start_ms;
+ absl::optional<size_t> outlier_frame_size;
+ absl::optional<int64_t> encode_start_ms;
uint8_t timing_flags = VideoSendTiming::kNotTriggered;
{
rtc::CritScope crit(&timing_params_lock_);
diff --git a/modules/video_coding/generic_encoder.h b/modules/video_coding/generic_encoder.h
index 03de626..0759f55 100644
--- a/modules/video_coding/generic_encoder.h
+++ b/modules/video_coding/generic_encoder.h
@@ -79,8 +79,8 @@
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,
- EncodedImage* encoded_image)
+ absl::optional<int64_t> ExtractEncodeStartTime(size_t simulcast_svc_idx,
+ EncodedImage* encoded_image)
RTC_EXCLUSIVE_LOCKS_REQUIRED(timing_params_lock_);
void FillTimingInfo(size_t simulcast_svc_idx, EncodedImage* encoded_image);
diff --git a/modules/video_coding/h264_sps_pps_tracker.cc b/modules/video_coding/h264_sps_pps_tracker.cc
index e8864e2..2b247c9 100644
--- a/modules/video_coding/h264_sps_pps_tracker.cc
+++ b/modules/video_coding/h264_sps_pps_tracker.cc
@@ -219,9 +219,9 @@
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) {
diff --git a/modules/video_coding/include/mock/mock_vcm_callbacks.h b/modules/video_coding/include/mock/mock_vcm_callbacks.h
index ab67f39..aab0e6a 100644
--- a/modules/video_coding/include/mock/mock_vcm_callbacks.h
+++ b/modules/video_coding/include/mock/mock_vcm_callbacks.h
@@ -34,7 +34,7 @@
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*));
diff --git a/modules/video_coding/include/mock/mock_video_codec_interface.h b/modules/video_coding/include/mock/mock_video_codec_interface.h
index 19ecbc4..4857c22 100644
--- a/modules/video_coding/include/mock/mock_video_codec_interface.h
+++ b/modules/video_coding/include/mock/mock_video_codec_interface.h
@@ -58,8 +58,8 @@
int64_t decode_time_ms));
MOCK_METHOD3(Decoded,
void(VideoFrame& decodedImage, // NOLINT
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp));
+ absl::optional<int32_t> decode_time_ms,
+ absl::optional<uint8_t> qp));
MOCK_METHOD1(ReceivedDecodedReferenceFrame,
int32_t(const uint64_t pictureId));
MOCK_METHOD1(ReceivedDecodedFrame, int32_t(const uint64_t pictureId));
diff --git a/modules/video_coding/include/video_coding_defines.h b/modules/video_coding/include/video_coding_defines.h
index 596e5ca..c24fdad 100644
--- a/modules/video_coding/include/video_coding_defines.h
+++ b/modules/video_coding/include/video_coding_defines.h
@@ -70,7 +70,7 @@
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);
diff --git a/modules/video_coding/packet_buffer.cc b/modules/video_coding/packet_buffer.cc
index ef48ef0..8960b1c 100644
--- a/modules/video_coding/packet_buffer.cc
+++ b/modules/video_coding/packet_buffer.cc
@@ -199,12 +199,12 @@
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_;
}
diff --git a/modules/video_coding/packet_buffer.h b/modules/video_coding/packet_buffer.h
index 9b1228d..51d2782 100644
--- a/modules/video_coding/packet_buffer.h
+++ b/modules/video_coding/packet_buffer.h
@@ -58,8 +58,8 @@
void PaddingReceived(uint16_t seq_num);
// Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
- rtc::Optional<int64_t> LastReceivedPacketMs() const;
- rtc::Optional<int64_t> LastReceivedKeyframePacketMs() const;
+ absl::optional<int64_t> LastReceivedPacketMs() const;
+ absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
// Returns number of different frames seen in the packet buffer
int GetUniqueFramesSeen() const;
@@ -159,13 +159,13 @@
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_);
- rtc::Optional<int64_t> last_received_keyframe_packet_ms_
+ absl::optional<int64_t> last_received_packet_ms_ RTC_GUARDED_BY(crit_);
+ 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_);
diff --git a/modules/video_coding/rtp_frame_reference_finder.cc b/modules/video_coding/rtp_frame_reference_finder.cc
index 8afdb02..9401243 100644
--- a/modules/video_coding/rtp_frame_reference_finder.cc
+++ b/modules/video_coding/rtp_frame_reference_finder.cc
@@ -240,7 +240,7 @@
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 @@
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.";
diff --git a/modules/video_coding/timing.cc b/modules/video_coding/timing.cc
index 40b8340..310467b 100644
--- a/modules/video_coding/timing.cc
+++ b/modules/video_coding/timing.cc
@@ -239,7 +239,7 @@
timing_frame_info_.emplace(info);
}
-rtc::Optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
+absl::optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
rtc::CritScope cs(&crit_sect_);
return timing_frame_info_;
}
diff --git a/modules/video_coding/timing.h b/modules/video_coding/timing.h
index 8c04ce2..08ec193 100644
--- a/modules/video_coding/timing.h
+++ b/modules/video_coding/timing.h
@@ -95,7 +95,7 @@
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 @@
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
diff --git a/modules/video_coding/utility/moving_average.cc b/modules/video_coding/utility/moving_average.cc
index 025899c..eb23e3d 100644
--- a/modules/video_coding/utility/moving_average.cc
+++ b/modules/video_coding/utility/moving_average.cc
@@ -23,13 +23,13 @@
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);
}
diff --git a/modules/video_coding/utility/moving_average.h b/modules/video_coding/utility/moving_average.h
index 0b7a700..77ce633 100644
--- a/modules/video_coding/utility/moving_average.h
+++ b/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 @@
explicit MovingAverage(size_t s);
~MovingAverage();
void AddSample(int sample);
- rtc::Optional<int> GetAverage() const;
- rtc::Optional<int> GetAverage(size_t num_samples) const;
+ absl::optional<int> GetAverage() const;
+ absl::optional<int> GetAverage(size_t num_samples) const;
void Reset();
size_t size() const;
diff --git a/modules/video_coding/utility/quality_scaler.cc b/modules/video_coding/utility/quality_scaler.cc
index 273d7a6..ea19e41 100644
--- a/modules/video_coding/utility/quality_scaler.cc
+++ b/modules/video_coding/utility/quality_scaler.cc
@@ -44,10 +44,10 @@
explicit QpSmoother(float alpha)
: alpha_(alpha), last_sample_ms_(rtc::TimeMillis()), smoother_(alpha) {}
- rtc::Optional<int> GetAvg() const {
+ absl::optional<int> GetAvg() const {
float value = smoother_.filtered();
if (value == rtc::ExpFilter::kValueUndefined) {
- return rtc::nullopt;
+ return absl::nullopt;
}
return static_cast<int>(value);
}
@@ -182,7 +182,7 @@
observed_enough_frames_ = true;
// Check if we should scale down due to high frame drop.
- const rtc::Optional<int> drop_rate =
+ const absl::optional<int> drop_rate =
config_.use_all_drop_reasons ? framedrop_percent_all_.GetAverage()
: framedrop_percent_media_opt_.GetAverage();
if (drop_rate && *drop_rate >= kFramedropPercentThreshold) {
@@ -192,10 +192,10 @@
}
// Check if we should scale up or down based on QP.
- const rtc::Optional<int> avg_qp_high = qp_smoother_high_
- ? qp_smoother_high_->GetAvg()
- : average_qp_.GetAverage();
- const rtc::Optional<int> avg_qp_low =
+ const absl::optional<int> avg_qp_high = qp_smoother_high_
+ ? qp_smoother_high_->GetAvg()
+ : average_qp_.GetAverage();
+ const absl::optional<int> avg_qp_low =
qp_smoother_low_ ? qp_smoother_low_->GetAvg() : average_qp_.GetAverage();
if (avg_qp_high && avg_qp_low) {
RTC_LOG(LS_INFO) << "Checking average QP " << *avg_qp_high << " ("
diff --git a/modules/video_coding/utility/quality_scaler.h b/modules/video_coding/utility/quality_scaler.h
index 996b9f2..3091ebf 100644
--- a/modules/video_coding/utility/quality_scaler.h
+++ b/modules/video_coding/utility/quality_scaler.h
@@ -14,7 +14,7 @@
#include <memory>
#include <utility>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/video_codecs/video_encoder.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/video_coding/utility/moving_average.h"
diff --git a/modules/video_coding/video_packet_buffer_unittest.cc b/modules/video_coding/video_packet_buffer_unittest.cc
index 84f5d2b..95bc171 100644
--- a/modules/video_coding/video_packet_buffer_unittest.cc
+++ b/modules/video_coding/video_packet_buffer_unittest.cc
@@ -711,8 +711,8 @@
}
TEST_F(TestPacketBuffer, PacketTimestamps) {
- rtc::Optional<int64_t> packet_ms;
- rtc::Optional<int64_t> packet_keyframe_ms;
+ absl::optional<int64_t> packet_ms;
+ absl::optional<int64_t> packet_keyframe_ms;
packet_ms = packet_buffer_->LastReceivedPacketMs();
packet_keyframe_ms = packet_buffer_->LastReceivedKeyframePacketMs();
