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 passing top level directories except rtc_base and api find $@ -type f \( -name \*.h -o -name \*.cc -o -name \*.mm \) \ -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: I9465c172e65ba6e6ed4e4fdc35b0b265038d6f71 Reviewed-on: https://webrtc-review.googlesource.com/84584 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23697}
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@ -171,8 +171,8 @@
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// Generate non-default certificate.
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if (keyType != rtc::KT_DEFAULT) {
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rtc::scoped_refptr<rtc::RTCCertificate> certificate =
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rtc::RTCCertificateGenerator::GenerateCertificate(
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rtc::KeyParams(keyType), rtc::Optional<uint64_t>());
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rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
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absl::optional<uint64_t>());
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if (!certificate) {
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RTCLogError(@"Failed to generate certificate.");
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return nullptr;
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@ -184,14 +184,13 @@
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nativeConfig->presume_writable_when_fully_relayed =
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_shouldPresumeWritableWhenFullyRelayed ? true : false;
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if (_iceCheckMinInterval != nil) {
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nativeConfig->ice_check_min_interval =
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rtc::Optional<int>(_iceCheckMinInterval.intValue);
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nativeConfig->ice_check_min_interval = absl::optional<int>(_iceCheckMinInterval.intValue);
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}
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if (_iceRegatherIntervalRange != nil) {
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std::unique_ptr<rtc::IntervalRange> nativeIntervalRange(
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_iceRegatherIntervalRange.nativeIntervalRange);
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nativeConfig->ice_regather_interval_range =
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rtc::Optional<rtc::IntervalRange>(*nativeIntervalRange);
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absl::optional<rtc::IntervalRange>(*nativeIntervalRange);
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}
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nativeConfig->sdp_semantics = [[self class] nativeSdpSemanticsForSdpSemantics:_sdpSemantics];
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if (_turnCustomizer) {
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@ -31,7 +31,7 @@
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if (self = [super init]) {
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self.hexString = hexString;
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rtc::Optional<webrtc::H264::ProfileLevelId> profile_level_id =
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absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
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webrtc::H264::ParseProfileLevelId([hexString cStringUsingEncoding:NSUTF8StringEncoding]);
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if (profile_level_id.has_value()) {
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self.profile = static_cast<RTCH264Profile>(profile_level_id->profile);
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@ -46,7 +46,7 @@
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self.profile = profile;
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self.level = level;
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rtc::Optional<std::string> hex_string =
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absl::optional<std::string> hex_string =
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webrtc::H264::ProfileLevelIdToString(webrtc::H264::ProfileLevelId(
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static_cast<webrtc::H264::Profile>(profile), static_cast<webrtc::H264::Level>(level)));
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self.hexString =
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@ -470,13 +470,13 @@ void PeerConnectionDelegateAdapter::OnAddTrack(
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maxBitrateBps:(nullable NSNumber *)maxBitrateBps {
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webrtc::PeerConnectionInterface::BitrateParameters params;
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if (minBitrateBps != nil) {
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params.min_bitrate_bps = rtc::Optional<int>(minBitrateBps.intValue);
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params.min_bitrate_bps = absl::optional<int>(minBitrateBps.intValue);
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}
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if (currentBitrateBps != nil) {
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params.current_bitrate_bps = rtc::Optional<int>(currentBitrateBps.intValue);
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params.current_bitrate_bps = absl::optional<int>(currentBitrateBps.intValue);
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}
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if (maxBitrateBps != nil) {
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params.max_bitrate_bps = rtc::Optional<int>(maxBitrateBps.intValue);
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params.max_bitrate_bps = absl::optional<int>(maxBitrateBps.intValue);
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}
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return _peerConnection->SetBitrate(params).ok();
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}
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@ -93,10 +93,10 @@ const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
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RTC_NOTREACHED();
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}
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if (_clockRate != nil) {
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parameters.clock_rate = rtc::Optional<int>(_clockRate.intValue);
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parameters.clock_rate = absl::optional<int>(_clockRate.intValue);
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}
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if (_numChannels != nil) {
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parameters.num_channels = rtc::Optional<int>(_numChannels.intValue);
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parameters.num_channels = absl::optional<int>(_numChannels.intValue);
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}
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for (NSString *paramKey in _parameters.allKeys) {
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std::string key = [NSString stdStringForString:paramKey];
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@ -44,13 +44,13 @@
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webrtc::RtpEncodingParameters parameters;
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parameters.active = _isActive;
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if (_maxBitrateBps != nil) {
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parameters.max_bitrate_bps = rtc::Optional<int>(_maxBitrateBps.intValue);
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parameters.max_bitrate_bps = absl::optional<int>(_maxBitrateBps.intValue);
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}
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if (_minBitrateBps != nil) {
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parameters.min_bitrate_bps = rtc::Optional<int>(_minBitrateBps.intValue);
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parameters.min_bitrate_bps = absl::optional<int>(_minBitrateBps.intValue);
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}
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if (_ssrc != nil) {
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parameters.ssrc = rtc::Optional<uint32_t>(_ssrc.unsignedLongValue);
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parameters.ssrc = absl::optional<uint32_t>(_ssrc.unsignedLongValue);
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}
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return parameters;
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}
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@ -43,9 +43,9 @@ namespace {
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using namespace webrtc::H264;
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NSString *MaxSupportedLevelForProfile(Profile profile) {
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const rtc::Optional<ProfileLevelId> profileLevelId = [UIDevice maxSupportedH264Profile];
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const absl::optional<ProfileLevelId> profileLevelId = [UIDevice maxSupportedH264Profile];
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if (profileLevelId && profileLevelId->profile >= profile) {
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const rtc::Optional<std::string> profileString =
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const absl::optional<std::string> profileString =
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ProfileLevelIdToString(ProfileLevelId(profile, profileLevelId->level));
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if (profileString) {
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return [NSString stringForStdString:*profileString];
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