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

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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}
This commit is contained in:
Danil Chapovalov
2018-06-21 13:32:56 +02:00
committed by Commit Bot
parent 1ff41eb784
commit 0bc58cf876
81 changed files with 371 additions and 365 deletions
Download Patch File Download Diff File Expand all files Collapse all files

30
api/peerconnectioninterface.h
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@ -361,7 +361,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
// 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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
// 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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
// 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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
// 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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
// 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