Replace rtc::Optional with absl::optional in audio, call and video

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

This CL generated by running script with parameters 'audio call video':
#!/bin/bash
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: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}

12 files changed, 53 insertions, 53 deletions

diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 75a62c2f75..709fd7fade 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -49,7 +49,6 @@ rtc_static_library("audio") {
     "../api:array_view",
     "../api:call_api",
     "../api:libjingle_peerconnection_api",
-    "../api:optional",
     "../api:transport_api",
     "../api/audio:aec3_factory",
     "../api/audio:audio_frame_api",
@@ -87,6 +86,7 @@ rtc_static_library("audio") {
     "../system_wrappers:field_trial_api",
     "../system_wrappers:metrics_api",
     "utility:audio_frame_operations",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 if (rtc_include_tests) {
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 941439fae1..b52563bed1 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -257,7 +257,7 @@ int AudioReceiveStream::id() const {
   return config_.rtp.remote_ssrc;
 }
 
-rtc::Optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
+absl::optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
   RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
   Syncable::Info info;
 
@@ -270,14 +270,14 @@ rtc::Optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
   if (!rtp_receiver->GetLatestTimestamps(
           &info.latest_received_capture_timestamp,
           &info.latest_receive_time_ms)) {
-    return rtc::nullopt;
+    return absl::nullopt;
   }
   if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs,
                           &info.capture_time_ntp_frac,
                           nullptr,
                           nullptr,
                           &info.capture_time_source_clock) != 0) {
-    return rtc::nullopt;
+    return absl::nullopt;
   }
 
   info.current_delay_ms = channel_proxy_->GetDelayEstimate();
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index 09007f0497..64552e35c0 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -80,7 +80,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
 
   // Syncable
   int id() const override;
-  rtc::Optional<Syncable::Info> GetInfo() const override;
+  absl::optional<Syncable::Info> GetInfo() const override;
   uint32_t GetPlayoutTimestamp() const override;
   void SetMinimumPlayoutDelay(int delay_ms) override;
 
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index fe955594ef..8ed78eb461 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -91,7 +91,7 @@ AudioSendStream::AudioSendStream(
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats,
-    const rtc::Optional<RtpState>& suspended_rtp_state,
+    const absl::optional<RtpState>& suspended_rtp_state,
     TimeInterval* overall_call_lifetime)
     : AudioSendStream(config,
                       audio_state,
@@ -115,7 +115,7 @@ AudioSendStream::AudioSendStream(
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats,
-    const rtc::Optional<RtpState>& suspended_rtp_state,
+    const absl::optional<RtpState>& suspended_rtp_state,
     TimeInterval* overall_call_lifetime,
     std::unique_ptr<voe::ChannelProxy> channel_proxy)
     : worker_queue_(worker_queue),
@@ -445,8 +445,8 @@ void AudioSendStream::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
 void AudioSendStream::OnPacketFeedbackVector(
     const std::vector<PacketFeedback>& packet_feedback_vector) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  rtc::Optional<float> plr;
-  rtc::Optional<float> rplr;
+  absl::optional<float> plr;
+  absl::optional<float> rplr;
   {
     rtc::CritScope lock(&packet_loss_tracker_cs_);
     packet_loss_tracker_.OnPacketFeedbackVector(packet_feedback_vector);
@@ -584,7 +584,7 @@ bool AudioSendStream::ReconfigureSendCodec(AudioSendStream* stream,
     return SetupSendCodec(stream, new_config);
   }
 
-  const rtc::Optional<int>& new_target_bitrate_bps =
+  const absl::optional<int>& new_target_bitrate_bps =
       new_config.send_codec_spec->target_bitrate_bps;
   // If a bitrate has been specified for the codec, use it over the
   // codec's default.
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index c51c7a3b2f..efc9c0e8fe 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -49,7 +49,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats,
-                  const rtc::Optional<RtpState>& suspended_rtp_state,
+                  const absl::optional<RtpState>& suspended_rtp_state,
                   TimeInterval* overall_call_lifetime);
   // For unit tests, which need to supply a mock channel proxy.
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
@@ -59,7 +59,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats,
-                  const rtc::Optional<RtpState>& suspended_rtp_state,
+                  const absl::optional<RtpState>& suspended_rtp_state,
                   TimeInterval* overall_call_lifetime,
                   std::unique_ptr<voe::ChannelProxy> channel_proxy);
   ~AudioSendStream() override;
@@ -146,7 +146,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
       RTC_GUARDED_BY(&packet_loss_tracker_cs_);
 
   RtpRtcp* rtp_rtcp_module_;
-  rtc::Optional<RtpState> const suspended_rtp_state_;
+  absl::optional<RtpState> const suspended_rtp_state_;
 
   std::unique_ptr<TimedTransport> timed_send_transport_adapter_;
   TimeInterval active_lifetime_;
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index d39912000c..d0bc45f4f7 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -109,17 +109,17 @@ rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() {
           std::begin(kCodecSpecs), std::end(kCodecSpecs))));
   ON_CALL(*factory.get(), QueryAudioEncoder(_))
       .WillByDefault(Invoke(
-          [](const SdpAudioFormat& format) -> rtc::Optional<AudioCodecInfo> {
+          [](const SdpAudioFormat& format) -> absl::optional<AudioCodecInfo> {
             for (const auto& spec : kCodecSpecs) {
               if (format == spec.format) {
                 return spec.info;
               }
             }
-            return rtc::nullopt;
+            return absl::nullopt;
           }));
   ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
       .WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format,
-                               rtc::Optional<AudioCodecPairId> codec_pair_id,
+                               absl::optional<AudioCodecPairId> codec_pair_id,
                                std::unique_ptr<AudioEncoder>* return_value) {
         *return_value = SetupAudioEncoderMock(payload_type, format);
       }));
@@ -166,7 +166,7 @@ struct ConfigHelper {
     return std::unique_ptr<internal::AudioSendStream>(
         new internal::AudioSendStream(
             stream_config_, audio_state_, &worker_queue_, &rtp_transport_,
-            &bitrate_allocator_, &event_log_, &rtcp_rtt_stats_, rtc::nullopt,
+            &bitrate_allocator_, &event_log_, &rtcp_rtt_stats_, absl::nullopt,
             &active_lifetime_,
             std::unique_ptr<voe::ChannelProxy>(channel_proxy_)));
   }
@@ -424,7 +424,7 @@ TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) {
   EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _))
       .WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString](
                            int payload_type, const SdpAudioFormat& format,
-                           rtc::Optional<AudioCodecPairId> codec_pair_id,
+                           absl::optional<AudioCodecPairId> codec_pair_id,
                            std::unique_ptr<AudioEncoder>* return_value) {
         auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
         EXPECT_CALL(*mock_encoder,
diff --git a/audio/channel.cc b/audio/channel.cc
index e5dc5c2814..ecde1725bd 100644
--- a/audio/channel.cc
+++ b/audio/channel.cc
@@ -509,7 +509,7 @@ Channel::Channel(rtc::TaskQueue* encoder_queue,
               0,
               false,
               rtc::scoped_refptr<AudioDecoderFactory>(),
-              rtc::nullopt) {
+              absl::nullopt) {
   RTC_DCHECK(encoder_queue);
   encoder_queue_ = encoder_queue;
 }
@@ -520,7 +520,7 @@ Channel::Channel(ProcessThread* module_process_thread,
                  size_t jitter_buffer_max_packets,
                  bool jitter_buffer_fast_playout,
                  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
-                 rtc::Optional<AudioCodecPairId> codec_pair_id)
+                 absl::optional<AudioCodecPairId> codec_pair_id)
     : event_log_proxy_(new RtcEventLogProxy()),
       rtp_payload_registry_(new RTPPayloadRegistry()),
       rtp_receive_statistics_(
diff --git a/audio/channel.h b/audio/channel.h
index 6f79868391..9830a2507d 100644
--- a/audio/channel.h
+++ b/audio/channel.h
@@ -16,11 +16,11 @@
 #include <string>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/call/audio_sink.h"
 #include "api/call/transport.h"
-#include "api/optional.h"
 #include "audio/audio_level.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/include/audio_coding_module.h"
@@ -158,7 +158,7 @@ class Channel
           size_t jitter_buffer_max_packets,
           bool jitter_buffer_fast_playout,
           rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
-          rtc::Optional<AudioCodecPairId> codec_pair_id);
+          absl::optional<AudioCodecPairId> codec_pair_id);
   virtual ~Channel();
 
   void SetSink(AudioSinkInterface* sink);
@@ -353,7 +353,7 @@ class Channel
   RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
 
   // Timestamp of the audio pulled from NetEq.
-  rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_;
+  absl::optional<uint32_t> jitter_buffer_playout_timestamp_;
 
   rtc::CriticalSection video_sync_lock_;
   uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_);
diff --git a/audio/time_interval.h b/audio/time_interval.h
index 88b2f7dd4f..79fe29d9d5 100644
--- a/audio/time_interval.h
+++ b/audio/time_interval.h
@@ -13,7 +13,7 @@
 
 #include <stdint.h>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 
 namespace webrtc {
 
@@ -57,7 +57,7 @@ class TimeInterval {
 
     int64_t first, last;
   };
-  rtc::Optional<Interval> interval_;
+  absl::optional<Interval> interval_;
 };
 
 }  // namespace webrtc
diff --git a/audio/transport_feedback_packet_loss_tracker.cc b/audio/transport_feedback_packet_loss_tracker.cc
index 101b6b4881..7e0c5c542f 100644
--- a/audio/transport_feedback_packet_loss_tracker.cc
+++ b/audio/transport_feedback_packet_loss_tracker.cc
@@ -116,12 +116,12 @@ void TransportFeedbackPacketLossTracker::OnPacketFeedbackVector(
   }
 }
 
-rtc::Optional<float>
-TransportFeedbackPacketLossTracker::GetPacketLossRate() const {
+absl::optional<float> TransportFeedbackPacketLossTracker::GetPacketLossRate()
+    const {
   return plr_state_.GetMetric();
 }
 
-rtc::Optional<float>
+absl::optional<float>
 TransportFeedbackPacketLossTracker::GetRecoverablePacketLossRate() const {
   return rplr_state_.GetMetric();
 }
@@ -344,20 +344,20 @@ void TransportFeedbackPacketLossTracker::Validate() const {  // Testing only!
   RTC_CHECK_EQ(rplr_state_.num_recoverable_losses_, recoverable_losses);
 }
 
-rtc::Optional<float>
-TransportFeedbackPacketLossTracker::PlrState::GetMetric() const {
+absl::optional<float> TransportFeedbackPacketLossTracker::PlrState::GetMetric()
+    const {
   const size_t total = num_lost_packets_ + num_received_packets_;
   if (total < min_num_acked_packets_) {
-    return rtc::nullopt;
+    return absl::nullopt;
   } else {
     return static_cast<float>(num_lost_packets_) / total;
   }
 }
 
-rtc::Optional<float>
-TransportFeedbackPacketLossTracker::RplrState::GetMetric() const {
+absl::optional<float> TransportFeedbackPacketLossTracker::RplrState::GetMetric()
+    const {
   if (num_acked_pairs_ < min_num_acked_pairs_) {
-    return rtc::nullopt;
+    return absl::nullopt;
   } else {
     return static_cast<float>(num_recoverable_losses_) / num_acked_pairs_;
   }
diff --git a/audio/transport_feedback_packet_loss_tracker.h b/audio/transport_feedback_packet_loss_tracker.h
index 4ad49024a8..7d58d6c4bd 100644
--- a/audio/transport_feedback_packet_loss_tracker.h
+++ b/audio/transport_feedback_packet_loss_tracker.h
@@ -14,7 +14,7 @@
 #include <map>
 #include <vector>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 
 namespace webrtc {
 
@@ -43,11 +43,11 @@ class TransportFeedbackPacketLossTracker final {
 
   // Returns the packet loss rate, if the window has enough packet statuses to
   // reliably compute it. Otherwise, returns empty.
-  rtc::Optional<float> GetPacketLossRate() const;
+  absl::optional<float> GetPacketLossRate() const;
 
   // Returns the first-order-FEC recoverable packet loss rate, if the window has
   // enough status pairs to reliably compute it. Otherwise, returns empty.
-  rtc::Optional<float> GetRecoverablePacketLossRate() const;
+  absl::optional<float> GetRecoverablePacketLossRate() const;
 
   // Verifies that the internal states are correct. Only used for tests.
   void Validate() const;
@@ -108,7 +108,7 @@ class TransportFeedbackPacketLossTracker final {
       num_received_packets_ = 0;
       num_lost_packets_ = 0;
     }
-    rtc::Optional<float> GetMetric() const;
+    absl::optional<float> GetMetric() const;
     const size_t min_num_acked_packets_;
     size_t num_received_packets_;
     size_t num_lost_packets_;
@@ -124,7 +124,7 @@ class TransportFeedbackPacketLossTracker final {
       num_acked_pairs_ = 0;
       num_recoverable_losses_ = 0;
     }
-    rtc::Optional<float> GetMetric() const;
+    absl::optional<float> GetMetric() const;
     // Recoverable packets are those which were lost, but immediately followed
     // by a properly received packet. If that second packet carried FEC,
     // the data from the former (lost) packet could be recovered.
diff --git a/audio/transport_feedback_packet_loss_tracker_unittest.cc b/audio/transport_feedback_packet_loss_tracker_unittest.cc
index 8f8fe05d7a..b190c62c05 100644
--- a/audio/transport_feedback_packet_loss_tracker_unittest.cc
+++ b/audio/transport_feedback_packet_loss_tracker_unittest.cc
@@ -94,18 +94,18 @@ class TransportFeedbackPacketLossTrackerTest
   // value is as expected.
   void ValidatePacketLossStatistics(
       const TransportFeedbackPacketLossTracker& tracker,
-      rtc::Optional<float> expected_plr,
-      rtc::Optional<float> expected_rplr) {
-    // TODO(eladalon): Comparing the rtc::Optional<float> directly would have
+      absl::optional<float> expected_plr,
+      absl::optional<float> expected_rplr) {
+    // TODO(eladalon): Comparing the absl::optional<float> directly would have
     // given concise code, but less readable error messages. If we modify
-    // the way rtc::Optional is printed, we can get rid of this.
-    rtc::Optional<float> plr = tracker.GetPacketLossRate();
+    // the way absl::optional is printed, we can get rid of this.
+    absl::optional<float> plr = tracker.GetPacketLossRate();
     EXPECT_EQ(static_cast<bool>(expected_plr), static_cast<bool>(plr));
     if (expected_plr && plr) {
       EXPECT_EQ(*expected_plr, *plr);
     }
 
-    rtc::Optional<float> rplr = tracker.GetRecoverablePacketLossRate();
+    absl::optional<float> rplr = tracker.GetRecoverablePacketLossRate();
     EXPECT_EQ(static_cast<bool>(expected_rplr), static_cast<bool>(rplr));
     if (expected_rplr && rplr) {
       EXPECT_EQ(*expected_rplr, *rplr);
@@ -127,7 +127,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, EmptyWindow) {
   TransportFeedbackPacketLossTracker tracker(kDefaultMaxWindowSizeMs, 5, 5);
 
   // PLR and RPLR reported as unknown before reception of first feedback.
-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
 }
 
 // A feedback received for an empty window has no effect.
@@ -136,7 +136,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, EmptyWindowFeedback) {
 
   // Feedback doesn't correspond to any packets - ignored.
   AddTransportFeedbackAndValidate(&tracker, base_, {true, false, true});
-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
 
   // After the packets are transmitted, acking them would have an effect.
   SendPackets(&tracker, base_, 3, kDefaultSendIntervalMs);
@@ -153,7 +153,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, PartiallyFilledWindow) {
   // Expected window contents: [] -> [1001].
   SendPackets(&tracker, base_, 3, kDefaultSendIntervalMs);
   AddTransportFeedbackAndValidate(&tracker, base_, {true, false, false, true});
-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
 }
 
 // Sanity check on minimum filled window - PLR known, RPLR unknown.
@@ -166,7 +166,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, PlrMinimumFilledWindow) {
   SendPackets(&tracker, base_, 5, kDefaultSendIntervalMs);
   AddTransportFeedbackAndValidate(&tracker, base_,
                                   {true, false, false, true, true});
-  ValidatePacketLossStatistics(tracker, 2.0f / 5.0f, rtc::nullopt);
+  ValidatePacketLossStatistics(tracker, 2.0f / 5.0f, absl::nullopt);
 }
 
 // Sanity check on minimum filled window - PLR unknown, RPLR known.
@@ -179,7 +179,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, RplrMinimumFilledWindow) {
   SendPackets(&tracker, base_, 5, kDefaultSendIntervalMs);
   AddTransportFeedbackAndValidate(&tracker, base_,
                                   {true, false, false, true, true});
-  ValidatePacketLossStatistics(tracker, rtc::nullopt, 1.0f / 4.0f);
+  ValidatePacketLossStatistics(tracker, absl::nullopt, 1.0f / 4.0f);
 }
 
 // If packets are sent close enough together that the clock reading for both
@@ -203,7 +203,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, ExtendWindow) {
   // Expected window contents: [] -> [10011].
   AddTransportFeedbackAndValidate(&tracker, base_,
                                   {true, false, false, true, true});
-  ValidatePacketLossStatistics(tracker, 2.0f / 5.0f, rtc::nullopt);
+  ValidatePacketLossStatistics(tracker, 2.0f / 5.0f, absl::nullopt);
 
   // Expected window contents: [10011] -> [1001110101].
   AddTransportFeedbackAndValidate(&tracker, base_ + 5,
@@ -520,7 +520,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, RepeatedSeqNumResetsWindow) {
   // A reset occurs.
   SendPackets(&tracker, {static_cast<uint16_t>(base_ + 2)},
               kDefaultSendIntervalMs);
-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
 }
 
 // The window is reset by the sending of a packet which is 0x8000 or more
@@ -539,7 +539,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest,
   // A reset occurs.
   SendPackets(&tracker, {static_cast<uint16_t>(base_ + 5 + 0x8000)},
               kDefaultSendIntervalMs);
-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
 }
 
 #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)