Replace rtc::Optional with absl::optional in api

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

This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests

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: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}
diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 683a597..cdc3266 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -361,7 +361,7 @@
     // The below fields correspond to constraints from the deprecated
     // constraints interface for constructing a PeerConnection.
     //
-    // rtc::Optional fields can be "missing", in which case the implementation
+    // absl::optional fields can be "missing", in which case the implementation
     // default will be used.
     //////////////////////////////////////////////////////////////////////////
 
@@ -396,15 +396,15 @@
     // Minimum bitrate at which screencast video tracks will be encoded at.
     // This means adding padding bits up to this bitrate, which can help
     // when switching from a static scene to one with motion.
-    rtc::Optional<int> screencast_min_bitrate;
+    absl::optional<int> screencast_min_bitrate;
 
     // Use new combined audio/video bandwidth estimation?
-    rtc::Optional<bool> combined_audio_video_bwe;
+    absl::optional<bool> combined_audio_video_bwe;
 
     // Can be used to disable DTLS-SRTP. This should never be done, but can be
     // useful for testing purposes, for example in setting up a loopback call
     // with a single PeerConnection.
-    rtc::Optional<bool> enable_dtls_srtp;
+    absl::optional<bool> enable_dtls_srtp;
 
     /////////////////////////////////////////////////
     // The below fields are not part of the standard.
@@ -504,29 +504,29 @@
     // 3) ice_check_min_interval defines the minimal interval (equivalently the
     // maximum rate) that overrides the above two intervals when either of them
     // is less.
-    rtc::Optional<int> ice_check_interval_strong_connectivity;
-    rtc::Optional<int> ice_check_interval_weak_connectivity;
-    rtc::Optional<int> ice_check_min_interval;
+    absl::optional<int> ice_check_interval_strong_connectivity;
+    absl::optional<int> ice_check_interval_weak_connectivity;
+    absl::optional<int> ice_check_min_interval;
 
     // The min time period for which a candidate pair must wait for response to
     // connectivity checks before it becomes unwritable. This parameter
     // overrides the default value in the ICE implementation if set.
-    rtc::Optional<int> ice_unwritable_timeout;
+    absl::optional<int> ice_unwritable_timeout;
 
     // The min number of connectivity checks that a candidate pair must sent
     // without receiving response before it becomes unwritable. This parameter
     // overrides the default value in the ICE implementation if set.
-    rtc::Optional<int> ice_unwritable_min_checks;
+    absl::optional<int> ice_unwritable_min_checks;
 
     // The interval in milliseconds at which STUN candidates will resend STUN
     // binding requests to keep NAT bindings open.
-    rtc::Optional<int> stun_candidate_keepalive_interval;
+    absl::optional<int> stun_candidate_keepalive_interval;
 
     // ICE Periodic Regathering
     // If set, WebRTC will periodically create and propose candidates without
     // starting a new ICE generation. The regathering happens continuously with
     // interval specified in milliseconds by the uniform distribution [a, b].
-    rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
+    absl::optional<rtc::IntervalRange> ice_regather_interval_range;
 
     // Optional TurnCustomizer.
     // With this class one can modify outgoing TURN messages.
@@ -538,7 +538,7 @@
     // A candidate pair on a preferred network has a higher precedence in ICE
     // than one on an un-preferred network, regardless of priority or network
     // cost.
-    rtc::Optional<rtc::AdapterType> network_preference;
+    absl::optional<rtc::AdapterType> network_preference;
 
     // Configure the SDP semantics used by this PeerConnection. Note that the
     // WebRTC 1.0 specification requires kUnifiedPlan semantics. The
@@ -979,9 +979,9 @@
 
   // 0 <= min <= current <= max should hold for set parameters.
   struct BitrateParameters {
-    rtc::Optional<int> min_bitrate_bps;
-    rtc::Optional<int> current_bitrate_bps;
-    rtc::Optional<int> max_bitrate_bps;
+    absl::optional<int> min_bitrate_bps;
+    absl::optional<int> current_bitrate_bps;
+    absl::optional<int> max_bitrate_bps;
   };
 
   // SetBitrate limits the bandwidth allocated for all RTP streams sent by