Replace rtc::Optional with absl::optional in pc
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'pc'
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: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12
Reviewed-on: https://webrtc-review.googlesource.com/84181
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23669}
This commit is contained in:
Danil Chapovalov
Commit Bot
parent
751a817044
commit
66cadcc6b9
@ -68,7 +68,6 @@ rtc_static_library("rtc_pc_base") {
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"../api:array_view",
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"../api:call_api",
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../api:ortc_api",
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"../api/video:video_frame",
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"../call:rtp_interfaces",
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@ -86,6 +85,7 @@ rtc_static_library("rtc_pc_base") {
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"../rtc_base:rtc_task_queue",
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"../rtc_base:stringutils",
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"../system_wrappers:metrics_api",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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if (rtc_build_libsrtp) {
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@ -193,7 +193,6 @@ rtc_static_library("peerconnection") {
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"../api:call_api",
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"../api:fec_controller_api",
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../api:rtc_stats_api",
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"../api/video:video_frame",
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"../api/video_codecs:video_codecs_api",
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@ -215,6 +214,7 @@ rtc_static_library("peerconnection") {
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"../stats",
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"../system_wrappers",
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"../system_wrappers:field_trial_api",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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@ -506,7 +506,6 @@ if (rtc_include_tests) {
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"../api:callfactory_api",
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"../api:fakemetricsobserver",
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"../api:libjingle_peerconnection_test_api",
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"../api:optional",
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"../api:rtc_stats_api",
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"../api/audio_codecs:audio_codecs_api",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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@ -538,6 +537,7 @@ if (rtc_include_tests) {
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"../system_wrappers:runtime_enabled_features_default",
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"../test:audio_codec_mocks",
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"../test:test_support",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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if (is_android) {
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@ -363,7 +363,7 @@ void BaseChannel::OnWritableState(bool writable) {
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}
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void BaseChannel::OnNetworkRouteChanged(
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rtc::Optional<rtc::NetworkRoute> network_route) {
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absl::optional<rtc::NetworkRoute> network_route) {
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RTC_DCHECK(network_thread_->IsCurrent());
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rtc::NetworkRoute new_route;
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if (network_route) {
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@ -218,7 +218,7 @@ class BaseChannel : public rtc::MessageHandler,
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// From RtpTransportInternal
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void OnWritableState(bool writable);
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void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route);
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void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);
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bool PacketIsRtcp(const rtc::PacketTransportInternal* transport,
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const char* data,
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@ -863,7 +863,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
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// The transport channel becomes disconnected.
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fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
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rtc::Optional<rtc::NetworkRoute>(network_route));
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absl::optional<rtc::NetworkRoute>(network_route));
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});
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WaitForThreads();
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EXPECT_EQ(1, media_channel1->num_network_route_changes());
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@ -880,7 +880,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
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// The transport channel becomes connected.
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fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
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rtc::Optional<rtc::NetworkRoute>(network_route));
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absl::optional<rtc::NetworkRoute>(network_route));
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});
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WaitForThreads();
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EXPECT_EQ(1, media_channel1->num_network_route_changes());
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@ -1348,7 +1348,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
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return channel1_->SetRemoteContent(&content, SdpType::kOffer, NULL);
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}
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webrtc::RtpParameters BitrateLimitedParameters(rtc::Optional<int> limit) {
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webrtc::RtpParameters BitrateLimitedParameters(absl::optional<int> limit) {
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webrtc::RtpParameters parameters;
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webrtc::RtpEncodingParameters encoding;
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encoding.max_bitrate_bps = std::move(limit);
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@ -1357,7 +1357,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
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}
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void VerifyMaxBitrate(const webrtc::RtpParameters& parameters,
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rtc::Optional<int> expected_bitrate) {
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absl::optional<int> expected_bitrate) {
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EXPECT_EQ(1UL, parameters.encodings.size());
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EXPECT_EQ(expected_bitrate, parameters.encodings[0].max_bitrate_bps);
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}
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@ -1368,7 +1368,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
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SdpType::kOffer, NULL));
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EXPECT_EQ(media_channel1_->max_bps(), -1);
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VerifyMaxBitrate(media_channel1_->GetRtpSendParameters(kSsrc1),
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rtc::nullopt);
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absl::nullopt);
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}
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// Test that when a channel gets new RtpTransport with a call to
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@ -88,8 +88,8 @@ class DtlsSrtpTransport : public SrtpTransport {
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cricket::DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;
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// The encrypted header extension IDs.
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rtc::Optional<std::vector<int>> send_extension_ids_;
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rtc::Optional<std::vector<int>> recv_extension_ids_;
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absl::optional<std::vector<int>> send_extension_ids_;
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absl::optional<std::vector<int>> recv_extension_ids_;
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bool active_reset_srtp_params_ = false;
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};
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@ -112,7 +112,7 @@ const char* SdpTypeToString(SdpType type) {
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return "";
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}
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rtc::Optional<SdpType> SdpTypeFromString(const std::string& type_str) {
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absl::optional<SdpType> SdpTypeFromString(const std::string& type_str) {
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if (type_str == SessionDescriptionInterface::kOffer) {
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return SdpType::kOffer;
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} else if (type_str == SessionDescriptionInterface::kPrAnswer) {
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@ -120,14 +120,14 @@ rtc::Optional<SdpType> SdpTypeFromString(const std::string& type_str) {
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} else if (type_str == SessionDescriptionInterface::kAnswer) {
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return SdpType::kAnswer;
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} else {
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return rtc::nullopt;
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return absl::nullopt;
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}
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}
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// TODO(steveanton): Remove this default implementation once Chromium has been
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// updated.
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SdpType SessionDescriptionInterface::GetType() const {
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rtc::Optional<SdpType> maybe_type = SdpTypeFromString(type());
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absl::optional<SdpType> maybe_type = SdpTypeFromString(type());
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if (maybe_type) {
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return *maybe_type;
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} else {
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@ -142,7 +142,7 @@ SdpType SessionDescriptionInterface::GetType() const {
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SessionDescriptionInterface* CreateSessionDescription(const std::string& type,
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const std::string& sdp,
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SdpParseError* error) {
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rtc::Optional<SdpType> maybe_type = SdpTypeFromString(type);
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absl::optional<SdpType> maybe_type = SdpTypeFromString(type);
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if (!maybe_type) {
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return nullptr;
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}
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@ -170,7 +170,7 @@ std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
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JsepSessionDescription::JsepSessionDescription(SdpType type) : type_(type) {}
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JsepSessionDescription::JsepSessionDescription(const std::string& type) {
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rtc::Optional<SdpType> maybe_type = SdpTypeFromString(type);
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absl::optional<SdpType> maybe_type = SdpTypeFromString(type);
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if (maybe_type) {
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type_ = *maybe_type;
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} else {
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@ -284,14 +284,14 @@ void JsepTransport::SetNeedsIceRestartFlag() {
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}
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}
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rtc::Optional<rtc::SSLRole> JsepTransport::GetDtlsRole() const {
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absl::optional<rtc::SSLRole> JsepTransport::GetDtlsRole() const {
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RTC_DCHECK(rtp_dtls_transport_);
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rtc::SSLRole dtls_role;
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if (!rtp_dtls_transport_->GetDtlsRole(&dtls_role)) {
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return rtc::Optional<rtc::SSLRole>();
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return absl::optional<rtc::SSLRole>();
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}
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return rtc::Optional<rtc::SSLRole>(dtls_role);
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return absl::optional<rtc::SSLRole>(dtls_role);
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}
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bool JsepTransport::GetStats(TransportStats* stats) {
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@ -357,7 +357,7 @@ void JsepTransport::SetRemoteIceParameters(
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webrtc::RTCError JsepTransport::SetNegotiatedDtlsParameters(
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DtlsTransportInternal* dtls_transport,
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rtc::Optional<rtc::SSLRole> dtls_role,
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absl::optional<rtc::SSLRole> dtls_role,
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rtc::SSLFingerprint* remote_fingerprint) {
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RTC_DCHECK(dtls_transport);
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// Set SSL role. Role must be set before fingerprint is applied, which
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@ -483,7 +483,7 @@ webrtc::RTCError JsepTransport::NegotiateAndSetDtlsParameters(
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"without applying any offer.");
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}
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std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint;
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rtc::Optional<rtc::SSLRole> negotiated_dtls_role;
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absl::optional<rtc::SSLRole> negotiated_dtls_role;
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rtc::SSLFingerprint* local_fp =
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local_description_->transport_desc.identity_fingerprint.get();
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@ -531,7 +531,7 @@ webrtc::RTCError JsepTransport::NegotiateDtlsRole(
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SdpType local_description_type,
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ConnectionRole local_connection_role,
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ConnectionRole remote_connection_role,
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rtc::Optional<rtc::SSLRole>* negotiated_dtls_role) {
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absl::optional<rtc::SSLRole>* negotiated_dtls_role) {
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// From RFC 4145, section-4.1, The following are the values that the
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// 'setup' attribute can take in an offer/answer exchange:
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// Offer Answer
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@ -16,9 +16,9 @@
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/candidate.h"
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#include "api/jsep.h"
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#include "api/optional.h"
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#include "p2p/base/dtlstransport.h"
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#include "p2p/base/p2pconstants.h"
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#include "p2p/base/transportinfo.h"
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@ -125,9 +125,9 @@ class JsepTransport : public sigslot::has_slots<> {
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// changed ufrag/password).
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bool needs_ice_restart() const { return needs_ice_restart_; }
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// Returns role if negotiated, or empty Optional if it hasn't been negotiated
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// yet.
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rtc::Optional<rtc::SSLRole> GetDtlsRole() const;
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// Returns role if negotiated, or empty absl::optional if it hasn't been
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// negotiated yet.
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absl::optional<rtc::SSLRole> GetDtlsRole() const;
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// TODO(deadbeef): Make this const. See comment in transportcontroller.h.
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bool GetStats(TransportStats* stats);
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@ -200,7 +200,7 @@ class JsepTransport : public sigslot::has_slots<> {
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webrtc::SdpType local_description_type,
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ConnectionRole local_connection_role,
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ConnectionRole remote_connection_role,
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rtc::Optional<rtc::SSLRole>* negotiated_dtls_role);
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absl::optional<rtc::SSLRole>* negotiated_dtls_role);
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// Pushes down the ICE parameters from the local description, such
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// as the ICE ufrag and pwd.
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@ -212,7 +212,7 @@ class JsepTransport : public sigslot::has_slots<> {
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// Pushes down the DTLS parameters obtained via negotiation.
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webrtc::RTCError SetNegotiatedDtlsParameters(
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DtlsTransportInternal* dtls_transport,
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rtc::Optional<rtc::SSLRole> dtls_role,
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absl::optional<rtc::SSLRole> dtls_role,
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rtc::SSLFingerprint* remote_fingerprint);
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bool GetTransportStats(DtlsTransportInternal* dtls_transport,
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@ -238,8 +238,8 @@ class JsepTransport : public sigslot::has_slots<> {
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RtcpMuxFilter rtcp_mux_negotiator_;
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// Cache the encrypted header extension IDs for SDES negoitation.
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rtc::Optional<std::vector<int>> send_extension_ids_;
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rtc::Optional<std::vector<int>> recv_extension_ids_;
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absl::optional<std::vector<int>> send_extension_ids_;
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absl::optional<std::vector<int>> recv_extension_ids_;
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RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransport);
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};
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@ -759,7 +759,7 @@ TEST_F(JsepTransport2Test, RemoteOfferWithCurrentNegotiatedDtlsRole) {
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.ok());
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// Sanity check that role was actually negotiated.
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rtc::Optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
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absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
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ASSERT_TRUE(role);
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EXPECT_EQ(rtc::SSL_CLIENT, *role);
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@ -804,7 +804,7 @@ TEST_F(JsepTransport2Test, RemoteOfferThatChangesNegotiatedDtlsRole) {
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.ok());
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// Sanity check that role was actually negotiated.
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rtc::Optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
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absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
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ASSERT_TRUE(role);
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EXPECT_EQ(rtc::SSL_CLIENT, *role);
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@ -849,7 +849,7 @@ TEST_F(JsepTransport2Test, DtlsSetupWithLegacyAsAnswerer) {
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->SetRemoteJsepTransportDescription(remote_desc, SdpType::kAnswer)
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.ok());
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rtc::Optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
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absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
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ASSERT_TRUE(role);
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// Since legacy answer ommitted setup atribute, and we offered actpass, we
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// should act as passive (server).
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@ -192,16 +192,16 @@ bool JsepTransportController::NeedsIceRestart(
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return transport->needs_ice_restart();
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}
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rtc::Optional<rtc::SSLRole> JsepTransportController::GetDtlsRole(
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absl::optional<rtc::SSLRole> JsepTransportController::GetDtlsRole(
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const std::string& mid) const {
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if (!network_thread_->IsCurrent()) {
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return network_thread_->Invoke<rtc::Optional<rtc::SSLRole>>(
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return network_thread_->Invoke<absl::optional<rtc::SSLRole>>(
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RTC_FROM_HERE, [&] { return GetDtlsRole(mid); });
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}
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const cricket::JsepTransport* t = GetJsepTransportForMid(mid);
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if (!t) {
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return rtc::Optional<rtc::SSLRole>();
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return absl::optional<rtc::SSLRole>();
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}
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return t->GetDtlsRole();
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}
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@ -146,7 +146,7 @@ class JsepTransportController : public sigslot::has_slots<>,
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std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
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const std::string& mid) const;
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// Get negotiated role, if one has been negotiated.
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rtc::Optional<rtc::SSLRole> GetDtlsRole(const std::string& mid) const;
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absl::optional<rtc::SSLRole> GetDtlsRole(const std::string& mid) const;
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// TODO(deadbeef): GetStats isn't const because all the way down to
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// OpenSSLStreamAdapter, GetSslCipherSuite and GetDtlsSrtpCryptoSuite are not
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@ -206,8 +206,8 @@ class JsepTransportController : public sigslot::has_slots<>,
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const std::vector<int>& encrypted_extension_ids,
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int rtp_abs_sendtime_extn_id);
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rtc::Optional<std::string> bundled_mid() const {
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rtc::Optional<std::string> bundled_mid;
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absl::optional<std::string> bundled_mid() const {
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absl::optional<std::string> bundled_mid;
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if (bundle_group_ && bundle_group_->FirstContentName()) {
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bundled_mid = *(bundle_group_->FirstContentName());
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}
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@ -312,9 +312,9 @@ class JsepTransportController : public sigslot::has_slots<>,
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Config config_;
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const cricket::SessionDescription* local_desc_ = nullptr;
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const cricket::SessionDescription* remote_desc_ = nullptr;
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rtc::Optional<bool> initial_offerer_;
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absl::optional<bool> initial_offerer_;
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rtc::Optional<cricket::ContentGroup> bundle_group_;
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absl::optional<cricket::ContentGroup> bundle_group_;
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cricket::IceConfig ice_config_;
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cricket::IceRole ice_role_ = cricket::ICEROLE_CONTROLLING;
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@ -524,7 +524,7 @@ TEST_F(JsepTransportControllerTest, GetDtlsRole) {
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->SetLocalDescription(SdpType::kOffer, offer_desc.get())
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.ok());
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rtc::Optional<rtc::SSLRole> role =
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absl::optional<rtc::SSLRole> role =
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transport_controller_->GetDtlsRole(kAudioMid1);
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// The DTLS role is not decided yet.
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EXPECT_FALSE(role);
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@ -33,5 +33,5 @@ TEST(LocalAudioSourceTest, InitWithAudioOptions) {
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TEST(LocalAudioSourceTest, InitWithNoOptions) {
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rtc::scoped_refptr<LocalAudioSource> source =
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LocalAudioSource::Create(nullptr);
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EXPECT_EQ(rtc::nullopt, source->options().highpass_filter);
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EXPECT_EQ(absl::nullopt, source->options().highpass_filter);
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}
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@ -18,8 +18,8 @@
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#include <unordered_map>
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#include <utility>
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#include "absl/types/optional.h"
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#include "api/cryptoparams.h"
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#include "api/optional.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "media/base/h264_profile_level_id.h"
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#include "media/base/mediaconstants.h"
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@ -609,11 +609,11 @@ std::string GetStreamIdsString(rtc::ArrayView<const std::string> stream_ids) {
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return output;
|
||||
}
|
||||
|
||||
rtc::Optional<int> RTCConfigurationToIceConfigOptionalInt(
|
||||
absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
|
||||
int rtc_configuration_parameter) {
|
||||
if (rtc_configuration_parameter ==
|
||||
webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
|
||||
return rtc::nullopt;
|
||||
return absl::nullopt;
|
||||
}
|
||||
return rtc_configuration_parameter;
|
||||
}
|
||||
@ -665,9 +665,9 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
|
||||
int max_ipv6_networks;
|
||||
bool disable_link_local_networks;
|
||||
bool enable_rtp_data_channel;
|
||||
rtc::Optional<int> screencast_min_bitrate;
|
||||
rtc::Optional<bool> combined_audio_video_bwe;
|
||||
rtc::Optional<bool> enable_dtls_srtp;
|
||||
absl::optional<int> screencast_min_bitrate;
|
||||
absl::optional<bool> combined_audio_video_bwe;
|
||||
absl::optional<bool> enable_dtls_srtp;
|
||||
TcpCandidatePolicy tcp_candidate_policy;
|
||||
CandidateNetworkPolicy candidate_network_policy;
|
||||
int audio_jitter_buffer_max_packets;
|
||||
@ -681,16 +681,16 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
|
||||
bool presume_writable_when_fully_relayed;
|
||||
bool enable_ice_renomination;
|
||||
bool redetermine_role_on_ice_restart;
|
||||
rtc::Optional<int> ice_check_interval_strong_connectivity;
|
||||
rtc::Optional<int> ice_check_interval_weak_connectivity;
|
||||
rtc::Optional<int> ice_check_min_interval;
|
||||
rtc::Optional<int> ice_unwritable_timeout;
|
||||
rtc::Optional<int> ice_unwritable_min_checks;
|
||||
rtc::Optional<int> stun_candidate_keepalive_interval;
|
||||
rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
|
||||
absl::optional<int> ice_check_interval_strong_connectivity;
|
||||
absl::optional<int> ice_check_interval_weak_connectivity;
|
||||
absl::optional<int> ice_check_min_interval;
|
||||
absl::optional<int> ice_unwritable_timeout;
|
||||
absl::optional<int> ice_unwritable_min_checks;
|
||||
absl::optional<int> stun_candidate_keepalive_interval;
|
||||
absl::optional<rtc::IntervalRange> ice_regather_interval_range;
|
||||
webrtc::TurnCustomizer* turn_customizer;
|
||||
SdpSemantics sdp_semantics;
|
||||
rtc::Optional<rtc::AdapterType> network_preference;
|
||||
absl::optional<rtc::AdapterType> network_preference;
|
||||
bool active_reset_srtp_params;
|
||||
};
|
||||
static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
|
||||
@ -2714,8 +2714,8 @@ PeerConnection::AssociateTransceiver(cricket::ContentSource source,
|
||||
if (old_transceiver) {
|
||||
RTC_LOG(LS_INFO) << "Dissociating transceiver for MID=" << old_mid
|
||||
<< " since the media section is being recycled.";
|
||||
old_transceiver->internal()->set_mid(rtc::nullopt);
|
||||
old_transceiver->internal()->set_mline_index(rtc::nullopt);
|
||||
old_transceiver->internal()->set_mid(absl::nullopt);
|
||||
old_transceiver->internal()->set_mline_index(absl::nullopt);
|
||||
}
|
||||
}
|
||||
const MediaContentDescription* media_desc = content.media_description();
|
||||
@ -3674,9 +3674,9 @@ void PeerConnection::GetOptionsForPlanBOffer(
|
||||
(offer_answer_options.offer_to_receive_video > 0);
|
||||
}
|
||||
|
||||
rtc::Optional<size_t> audio_index;
|
||||
rtc::Optional<size_t> video_index;
|
||||
rtc::Optional<size_t> data_index;
|
||||
absl::optional<size_t> audio_index;
|
||||
absl::optional<size_t> video_index;
|
||||
absl::optional<size_t> data_index;
|
||||
// If a current description exists, generate m= sections in the same order,
|
||||
// using the first audio/video/data section that appears and rejecting
|
||||
// extraneous ones.
|
||||
@ -3922,9 +3922,9 @@ void PeerConnection::GetOptionsForPlanBAnswer(
|
||||
recv_video = (offer_answer_options.offer_to_receive_video > 0);
|
||||
}
|
||||
|
||||
rtc::Optional<size_t> audio_index;
|
||||
rtc::Optional<size_t> video_index;
|
||||
rtc::Optional<size_t> data_index;
|
||||
absl::optional<size_t> audio_index;
|
||||
absl::optional<size_t> video_index;
|
||||
absl::optional<size_t> data_index;
|
||||
|
||||
// Generate m= sections that match those in the offer.
|
||||
// Note that mediasession.cc will handle intersection our preferred
|
||||
@ -3983,9 +3983,9 @@ void PeerConnection::GenerateMediaDescriptionOptions(
|
||||
const SessionDescriptionInterface* session_desc,
|
||||
RtpTransceiverDirection audio_direction,
|
||||
RtpTransceiverDirection video_direction,
|
||||
rtc::Optional<size_t>* audio_index,
|
||||
rtc::Optional<size_t>* video_index,
|
||||
rtc::Optional<size_t>* data_index,
|
||||
absl::optional<size_t>* audio_index,
|
||||
absl::optional<size_t>* video_index,
|
||||
absl::optional<size_t>* data_index,
|
||||
cricket::MediaSessionOptions* session_options) {
|
||||
for (const cricket::ContentInfo& content :
|
||||
session_desc->description()->contents()) {
|
||||
@ -4054,17 +4054,17 @@ PeerConnection::GetMediaDescriptionOptionsForRejectedData(
|
||||
return options;
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> PeerConnection::GetDataMid() const {
|
||||
absl::optional<std::string> PeerConnection::GetDataMid() const {
|
||||
switch (data_channel_type_) {
|
||||
case cricket::DCT_RTP:
|
||||
if (!rtp_data_channel_) {
|
||||
return rtc::nullopt;
|
||||
return absl::nullopt;
|
||||
}
|
||||
return rtp_data_channel_->content_name();
|
||||
case cricket::DCT_SCTP:
|
||||
return sctp_mid_;
|
||||
default:
|
||||
return rtc::nullopt;
|
||||
return absl::nullopt;
|
||||
}
|
||||
}
|
||||
|
||||
@ -4727,7 +4727,7 @@ bool PeerConnection::ReconfigurePortAllocator_n(
|
||||
int candidate_pool_size,
|
||||
bool prune_turn_ports,
|
||||
webrtc::TurnCustomizer* turn_customizer,
|
||||
rtc::Optional<int> stun_candidate_keepalive_interval) {
|
||||
absl::optional<int> stun_candidate_keepalive_interval) {
|
||||
port_allocator_->set_candidate_filter(
|
||||
ConvertIceTransportTypeToCandidateFilter(type));
|
||||
// According to JSEP, after setLocalDescription, changing the candidate pool
|
||||
@ -5123,15 +5123,15 @@ bool PeerConnection::ReadyToSendData() const {
|
||||
sctp_ready_to_send_data_;
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> PeerConnection::sctp_transport_name() const {
|
||||
absl::optional<std::string> PeerConnection::sctp_transport_name() const {
|
||||
if (sctp_mid_ && transport_controller_) {
|
||||
auto dtls_transport = transport_controller_->GetDtlsTransport(*sctp_mid_);
|
||||
if (dtls_transport) {
|
||||
return dtls_transport->transport_name();
|
||||
}
|
||||
return rtc::Optional<std::string>();
|
||||
return absl::optional<std::string>();
|
||||
}
|
||||
return rtc::Optional<std::string>();
|
||||
return absl::optional<std::string>();
|
||||
}
|
||||
|
||||
cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
|
||||
@ -5158,7 +5158,7 @@ std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
|
||||
rtp_data_channel_->transport_name();
|
||||
}
|
||||
if (sctp_transport_) {
|
||||
rtc::Optional<std::string> transport_name = sctp_transport_name();
|
||||
absl::optional<std::string> transport_name = sctp_transport_name();
|
||||
RTC_DCHECK(transport_name);
|
||||
transport_names_by_mid[*sctp_mid_] = *transport_name;
|
||||
}
|
||||
@ -6036,7 +6036,7 @@ void PeerConnection::ReportTransportStats() {
|
||||
cricket::MEDIA_TYPE_DATA);
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> transport_name = sctp_transport_name();
|
||||
absl::optional<std::string> transport_name = sctp_transport_name();
|
||||
if (transport_name) {
|
||||
media_types_by_transport_name[*transport_name].insert(
|
||||
cricket::MEDIA_TYPE_DATA);
|
||||
|
||||
@ -247,11 +247,11 @@ class PeerConnection : public PeerConnectionInternal,
|
||||
return sctp_data_channels_;
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> sctp_content_name() const override {
|
||||
absl::optional<std::string> sctp_content_name() const override {
|
||||
return sctp_mid_;
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> sctp_transport_name() const override;
|
||||
absl::optional<std::string> sctp_transport_name() const override;
|
||||
|
||||
cricket::CandidateStatsList GetPooledCandidateStats() const override;
|
||||
std::map<std::string, std::string> GetTransportNamesByMid() const override;
|
||||
@ -516,9 +516,9 @@ class PeerConnection : public PeerConnectionInternal,
|
||||
const SessionDescriptionInterface* session_desc,
|
||||
RtpTransceiverDirection audio_direction,
|
||||
RtpTransceiverDirection video_direction,
|
||||
rtc::Optional<size_t>* audio_index,
|
||||
rtc::Optional<size_t>* video_index,
|
||||
rtc::Optional<size_t>* data_index,
|
||||
absl::optional<size_t>* audio_index,
|
||||
absl::optional<size_t>* video_index,
|
||||
absl::optional<size_t>* data_index,
|
||||
cricket::MediaSessionOptions* session_options);
|
||||
|
||||
// Generates the active MediaDescriptionOptions for the local data channel
|
||||
@ -534,7 +534,7 @@ class PeerConnection : public PeerConnectionInternal,
|
||||
// Returns the MID for the data section associated with either the
|
||||
// RtpDataChannel or SCTP data channel, if it has been set. If no data
|
||||
// channels are configured this will return nullopt.
|
||||
rtc::Optional<std::string> GetDataMid() const;
|
||||
absl::optional<std::string> GetDataMid() const;
|
||||
|
||||
// Remove all local and remote senders of type |media_type|.
|
||||
// Called when a media type is rejected (m-line set to port 0).
|
||||
@ -665,7 +665,7 @@ class PeerConnection : public PeerConnectionInternal,
|
||||
int candidate_pool_size,
|
||||
bool prune_turn_ports,
|
||||
webrtc::TurnCustomizer* turn_customizer,
|
||||
rtc::Optional<int> stun_candidate_keepalive_interval);
|
||||
absl::optional<int> stun_candidate_keepalive_interval);
|
||||
|
||||
void SetMetricObserver_n(UMAObserver* observer);
|
||||
|
||||
@ -976,7 +976,7 @@ class PeerConnection : public PeerConnectionInternal,
|
||||
|
||||
std::unique_ptr<cricket::SctpTransportInternal> sctp_transport_;
|
||||
// |sctp_mid_| is the content name (MID) in SDP.
|
||||
rtc::Optional<std::string> sctp_mid_;
|
||||
absl::optional<std::string> sctp_mid_;
|
||||
// Value cached on signaling thread. Only updated when SctpReadyToSendData
|
||||
// fires on the signaling thread.
|
||||
bool sctp_ready_to_send_data_ = false;
|
||||
|
||||
@ -66,11 +66,11 @@ class PeerConnectionWrapperForDataChannelTest : public PeerConnectionWrapper {
|
||||
sctp_transport_factory_ = sctp_transport_factory;
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> sctp_content_name() {
|
||||
absl::optional<std::string> sctp_content_name() {
|
||||
return GetInternalPeerConnection()->sctp_content_name();
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> sctp_transport_name() {
|
||||
absl::optional<std::string> sctp_transport_name() {
|
||||
return GetInternalPeerConnection()->sctp_transport_name();
|
||||
}
|
||||
|
||||
|
||||
@ -981,7 +981,7 @@ TEST_F(PeerConnectionIceConfigTest, SetStunCandidateKeepaliveInterval) {
|
||||
config.ice_candidate_pool_size = 1;
|
||||
CreatePeerConnection(config);
|
||||
ASSERT_NE(port_allocator_, nullptr);
|
||||
rtc::Optional<int> actual_stun_keepalive_interval =
|
||||
absl::optional<int> actual_stun_keepalive_interval =
|
||||
port_allocator_->stun_candidate_keepalive_interval();
|
||||
EXPECT_EQ(actual_stun_keepalive_interval.value_or(-1), 123);
|
||||
config.stun_candidate_keepalive_interval = 321;
|
||||
|
||||
@ -304,7 +304,7 @@ TEST_F(PeerConnectionJsepTest,
|
||||
|
||||
auto transceivers = callee->pc()->GetTransceivers();
|
||||
ASSERT_EQ(2u, transceivers.size());
|
||||
EXPECT_EQ(rtc::nullopt, transceivers[0]->mid());
|
||||
EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
|
||||
EXPECT_EQ(caller_audio->mid(), transceivers[1]->mid());
|
||||
}
|
||||
|
||||
@ -322,7 +322,7 @@ TEST_F(PeerConnectionJsepTest,
|
||||
|
||||
auto transceivers = callee->pc()->GetTransceivers();
|
||||
ASSERT_EQ(2u, transceivers.size());
|
||||
EXPECT_EQ(rtc::nullopt, transceivers[0]->mid());
|
||||
EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
|
||||
EXPECT_EQ(caller->pc()->GetTransceivers()[0]->mid(), transceivers[1]->mid());
|
||||
EXPECT_EQ(MediaStreamTrackInterface::kAudioKind,
|
||||
transceivers[1]->receiver()->track()->kind());
|
||||
@ -341,7 +341,7 @@ TEST_F(PeerConnectionJsepTest,
|
||||
|
||||
auto transceivers = callee->pc()->GetTransceivers();
|
||||
ASSERT_EQ(2u, transceivers.size());
|
||||
EXPECT_EQ(rtc::nullopt, transceivers[0]->mid());
|
||||
EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
|
||||
EXPECT_EQ(caller->pc()->GetTransceivers()[0]->mid(), transceivers[1]->mid());
|
||||
EXPECT_EQ(MediaStreamTrackInterface::kAudioKind,
|
||||
transceivers[1]->receiver()->track()->kind());
|
||||
@ -360,7 +360,7 @@ TEST_F(PeerConnectionJsepTest, SetRemoteOfferDoesNotReuseStoppedTransceiver) {
|
||||
|
||||
auto transceivers = callee->pc()->GetTransceivers();
|
||||
ASSERT_EQ(2u, transceivers.size());
|
||||
EXPECT_EQ(rtc::nullopt, transceivers[0]->mid());
|
||||
EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
|
||||
EXPECT_TRUE(transceivers[0]->stopped());
|
||||
EXPECT_EQ(caller->pc()->GetTransceivers()[0]->mid(), transceivers[1]->mid());
|
||||
EXPECT_FALSE(transceivers[1]->stopped());
|
||||
@ -606,7 +606,7 @@ TEST_F(PeerConnectionJsepTest,
|
||||
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
|
||||
EXPECT_TRUE(first_transceiver->stopped());
|
||||
// First transceivers aren't dissociated yet.
|
||||
ASSERT_NE(rtc::nullopt, first_transceiver->mid());
|
||||
ASSERT_NE(absl::nullopt, first_transceiver->mid());
|
||||
std::string first_mid = *first_transceiver->mid();
|
||||
EXPECT_EQ(first_mid, callee->pc()->GetTransceivers()[0]->mid());
|
||||
|
||||
@ -625,10 +625,10 @@ TEST_F(PeerConnectionJsepTest,
|
||||
// associate the new transceivers.
|
||||
ASSERT_TRUE(
|
||||
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
|
||||
EXPECT_EQ(rtc::nullopt, first_transceiver->mid());
|
||||
EXPECT_EQ(absl::nullopt, first_transceiver->mid());
|
||||
EXPECT_EQ(second_mid, caller->pc()->GetTransceivers()[1]->mid());
|
||||
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
|
||||
EXPECT_EQ(rtc::nullopt, callee->pc()->GetTransceivers()[0]->mid());
|
||||
EXPECT_EQ(absl::nullopt, callee->pc()->GetTransceivers()[0]->mid());
|
||||
EXPECT_EQ(second_mid, callee->pc()->GetTransceivers()[1]->mid());
|
||||
|
||||
// The new answer should also recycle the m section correctly.
|
||||
@ -644,11 +644,11 @@ TEST_F(PeerConnectionJsepTest,
|
||||
ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
|
||||
auto caller_transceivers = caller->pc()->GetTransceivers();
|
||||
ASSERT_EQ(2u, caller_transceivers.size());
|
||||
EXPECT_EQ(rtc::nullopt, caller_transceivers[0]->mid());
|
||||
EXPECT_EQ(absl::nullopt, caller_transceivers[0]->mid());
|
||||
EXPECT_EQ(second_mid, caller_transceivers[1]->mid());
|
||||
auto callee_transceivers = callee->pc()->GetTransceivers();
|
||||
ASSERT_EQ(2u, callee_transceivers.size());
|
||||
EXPECT_EQ(rtc::nullopt, callee_transceivers[0]->mid());
|
||||
EXPECT_EQ(absl::nullopt, callee_transceivers[0]->mid());
|
||||
EXPECT_EQ(second_mid, callee_transceivers[1]->mid());
|
||||
}
|
||||
|
||||
@ -687,7 +687,7 @@ TEST_F(PeerConnectionJsepTest, CreateOfferRecyclesWhenOfferingTwice) {
|
||||
// Make sure that the caller's transceivers are associated correctly.
|
||||
auto caller_transceivers = caller->pc()->GetTransceivers();
|
||||
ASSERT_EQ(2u, caller_transceivers.size());
|
||||
EXPECT_EQ(rtc::nullopt, caller_transceivers[0]->mid());
|
||||
EXPECT_EQ(absl::nullopt, caller_transceivers[0]->mid());
|
||||
EXPECT_EQ(second_mid, caller_transceivers[1]->mid());
|
||||
EXPECT_FALSE(caller_transceivers[1]->stopped());
|
||||
}
|
||||
@ -737,7 +737,7 @@ TEST_P(RecycleMediaSectionTest, VerifyOfferAnswerAndTransceivers) {
|
||||
// the MID for the new transceiver.
|
||||
ASSERT_TRUE(
|
||||
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
|
||||
EXPECT_EQ(rtc::nullopt, first_transceiver->mid());
|
||||
EXPECT_EQ(absl::nullopt, first_transceiver->mid());
|
||||
EXPECT_EQ(second_mid, second_transceiver->mid());
|
||||
|
||||
// Setting the remote offer will dissociate the previous transceiver and
|
||||
@ -745,7 +745,7 @@ TEST_P(RecycleMediaSectionTest, VerifyOfferAnswerAndTransceivers) {
|
||||
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
|
||||
auto callee_transceivers = callee->pc()->GetTransceivers();
|
||||
ASSERT_EQ(2u, callee_transceivers.size());
|
||||
EXPECT_EQ(rtc::nullopt, callee_transceivers[0]->mid());
|
||||
EXPECT_EQ(absl::nullopt, callee_transceivers[0]->mid());
|
||||
EXPECT_EQ(first_type_, callee_transceivers[0]->media_type());
|
||||
EXPECT_EQ(second_mid, callee_transceivers[1]->mid());
|
||||
EXPECT_EQ(second_type_, callee_transceivers[1]->media_type());
|
||||
|
||||
@ -791,10 +791,10 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan,
|
||||
auto caller = CreatePeerConnection();
|
||||
|
||||
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
||||
EXPECT_EQ(rtc::nullopt, transceiver->mid());
|
||||
EXPECT_EQ(absl::nullopt, transceiver->mid());
|
||||
EXPECT_FALSE(transceiver->stopped());
|
||||
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
|
||||
EXPECT_EQ(rtc::nullopt, transceiver->current_direction());
|
||||
EXPECT_EQ(absl::nullopt, transceiver->current_direction());
|
||||
}
|
||||
|
||||
// Test that adding a transceiver with the audio kind creates an audio sender
|
||||
|
||||
@ -270,7 +270,7 @@ CreateForwardingMockDecoderFactory(
|
||||
.WillRepeatedly(
|
||||
Invoke([real_decoder_factory](
|
||||
const webrtc::SdpAudioFormat& format,
|
||||
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id,
|
||||
absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
|
||||
std::unique_ptr<webrtc::AudioDecoder>* return_value) {
|
||||
auto real_decoder =
|
||||
real_decoder_factory->MakeAudioDecoder(format, codec_pair_id);
|
||||
@ -284,7 +284,7 @@ CreateForwardingMockDecoderFactory(
|
||||
|
||||
struct AudioEncoderUnicornSparklesRainbow {
|
||||
using Config = webrtc::AudioEncoderL16::Config;
|
||||
static rtc::Optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
|
||||
static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
|
||||
if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
|
||||
const webrtc::SdpAudioFormat::Parameters expected_params = {
|
||||
{"num_horns", "1"}};
|
||||
@ -293,7 +293,7 @@ struct AudioEncoderUnicornSparklesRainbow {
|
||||
format.name = "L16";
|
||||
return webrtc::AudioEncoderL16::SdpToConfig(format);
|
||||
} else {
|
||||
return rtc::nullopt;
|
||||
return absl::nullopt;
|
||||
}
|
||||
}
|
||||
static void AppendSupportedEncoders(
|
||||
@ -313,7 +313,7 @@ struct AudioEncoderUnicornSparklesRainbow {
|
||||
static std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
|
||||
const Config& config,
|
||||
int payload_type,
|
||||
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id = rtc::nullopt) {
|
||||
absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
|
||||
return webrtc::AudioEncoderL16::MakeAudioEncoder(config, payload_type,
|
||||
codec_pair_id);
|
||||
}
|
||||
@ -321,7 +321,7 @@ struct AudioEncoderUnicornSparklesRainbow {
|
||||
|
||||
struct AudioDecoderUnicornSparklesRainbow {
|
||||
using Config = webrtc::AudioDecoderL16::Config;
|
||||
static rtc::Optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
|
||||
static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
|
||||
if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
|
||||
const webrtc::SdpAudioFormat::Parameters expected_params = {
|
||||
{"num_horns", "1"}};
|
||||
@ -330,7 +330,7 @@ struct AudioDecoderUnicornSparklesRainbow {
|
||||
format.name = "L16";
|
||||
return webrtc::AudioDecoderL16::SdpToConfig(format);
|
||||
} else {
|
||||
return rtc::nullopt;
|
||||
return absl::nullopt;
|
||||
}
|
||||
}
|
||||
static void AppendSupportedDecoders(
|
||||
@ -346,7 +346,7 @@ struct AudioDecoderUnicornSparklesRainbow {
|
||||
}
|
||||
static std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
|
||||
const Config& config,
|
||||
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id = rtc::nullopt) {
|
||||
absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
|
||||
return webrtc::AudioDecoderL16::MakeAudioDecoder(config, codec_pair_id);
|
||||
}
|
||||
};
|
||||
@ -392,14 +392,14 @@ TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
|
||||
std::vector<webrtc::AudioCodecSpec> GetSupportedEncoders() override {
|
||||
return fact_->GetSupportedEncoders();
|
||||
}
|
||||
rtc::Optional<webrtc::AudioCodecInfo> QueryAudioEncoder(
|
||||
absl::optional<webrtc::AudioCodecInfo> QueryAudioEncoder(
|
||||
const webrtc::SdpAudioFormat& format) override {
|
||||
return fact_->QueryAudioEncoder(format);
|
||||
}
|
||||
std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
|
||||
int payload_type,
|
||||
const webrtc::SdpAudioFormat& format,
|
||||
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id) override {
|
||||
absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
|
||||
EXPECT_TRUE(codec_pair_id.has_value());
|
||||
codec_ids_->push_back(*codec_pair_id);
|
||||
return fact_->MakeAudioEncoder(payload_type, format, codec_pair_id);
|
||||
@ -424,7 +424,7 @@ TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
|
||||
}
|
||||
std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
|
||||
const webrtc::SdpAudioFormat& format,
|
||||
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id) override {
|
||||
absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
|
||||
EXPECT_TRUE(codec_pair_id.has_value());
|
||||
codec_ids_->push_back(*codec_pair_id);
|
||||
return fact_->MakeAudioDecoder(format, codec_pair_id);
|
||||
|
||||
@ -2487,7 +2487,7 @@ TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
|
||||
// require a very complex set of mocks.
|
||||
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) {
|
||||
PeerConnectionInterface::RTCConfiguration config;
|
||||
config.ice_check_min_interval = rtc::nullopt;
|
||||
config.ice_check_min_interval = absl::nullopt;
|
||||
CreatePeerConnection(config, nullptr);
|
||||
config = pc_->GetConfiguration();
|
||||
config.ice_check_min_interval = 100;
|
||||
|
||||
@ -53,8 +53,8 @@ class PeerConnectionInternal : public PeerConnectionInterface {
|
||||
virtual std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels()
|
||||
const = 0;
|
||||
|
||||
virtual rtc::Optional<std::string> sctp_content_name() const = 0;
|
||||
virtual rtc::Optional<std::string> sctp_transport_name() const = 0;
|
||||
virtual absl::optional<std::string> sctp_content_name() const = 0;
|
||||
virtual absl::optional<std::string> sctp_transport_name() const = 0;
|
||||
|
||||
virtual cricket::CandidateStatsList GetPooledCandidateStats() const = 0;
|
||||
|
||||
|
||||
@ -17,7 +17,7 @@
|
||||
#include <string>
|
||||
#include <vector>
|
||||
|
||||
#include "api/optional.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/stats/rtcstats_objects.h"
|
||||
#include "api/stats/rtcstatscollectorcallback.h"
|
||||
#include "api/stats/rtcstatsreport.h"
|
||||
@ -145,8 +145,8 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface,
|
||||
struct RtpTransceiverStatsInfo {
|
||||
rtc::scoped_refptr<RtpTransceiver> transceiver;
|
||||
cricket::MediaType media_type;
|
||||
rtc::Optional<std::string> mid;
|
||||
rtc::Optional<std::string> transport_name;
|
||||
absl::optional<std::string> mid;
|
||||
absl::optional<std::string> transport_name;
|
||||
std::unique_ptr<TrackMediaInfoMap> track_media_info_map;
|
||||
};
|
||||
|
||||
|
||||
@ -1096,7 +1096,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
|
||||
connection_info.sent_total_bytes = 42;
|
||||
connection_info.recv_total_bytes = 1234;
|
||||
connection_info.total_round_trip_time_ms = 0;
|
||||
connection_info.current_round_trip_time_ms = rtc::nullopt;
|
||||
connection_info.current_round_trip_time_ms = absl::nullopt;
|
||||
connection_info.recv_ping_requests = 2020;
|
||||
connection_info.sent_ping_requests_total = 2020;
|
||||
connection_info.sent_ping_requests_before_first_response = 2000;
|
||||
@ -1638,7 +1638,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
|
||||
video_media_info.receivers[0].plis_sent = 6;
|
||||
video_media_info.receivers[0].nacks_sent = 7;
|
||||
video_media_info.receivers[0].frames_decoded = 8;
|
||||
video_media_info.receivers[0].qp_sum = rtc::nullopt;
|
||||
video_media_info.receivers[0].qp_sum = absl::nullopt;
|
||||
|
||||
RtpCodecParameters codec_parameters;
|
||||
codec_parameters.payload_type = 42;
|
||||
@ -1757,7 +1757,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
|
||||
video_media_info.senders[0].bytes_sent = 6;
|
||||
video_media_info.senders[0].codec_payload_type = 42;
|
||||
video_media_info.senders[0].frames_encoded = 8;
|
||||
video_media_info.senders[0].qp_sum = rtc::nullopt;
|
||||
video_media_info.senders[0].qp_sum = absl::nullopt;
|
||||
|
||||
RtpCodecParameters codec_parameters;
|
||||
codec_parameters.payload_type = 42;
|
||||
|
||||
@ -137,7 +137,7 @@ class AudioRtpReceiver : public ObserverInterface,
|
||||
const rtc::scoped_refptr<RemoteAudioSource> source_;
|
||||
const rtc::scoped_refptr<AudioTrackInterface> track_;
|
||||
cricket::VoiceMediaChannel* media_channel_ = nullptr;
|
||||
rtc::Optional<uint32_t> ssrc_;
|
||||
absl::optional<uint32_t> ssrc_;
|
||||
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
|
||||
bool cached_track_enabled_;
|
||||
double cached_volume_ = 1;
|
||||
@ -223,7 +223,7 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
|
||||
rtc::Thread* const worker_thread_;
|
||||
const std::string id_;
|
||||
cricket::VideoMediaChannel* media_channel_ = nullptr;
|
||||
rtc::Optional<uint32_t> ssrc_;
|
||||
absl::optional<uint32_t> ssrc_;
|
||||
// |source_| is held here to be able to change the state of the source when
|
||||
// the VideoRtpReceiver is stopped.
|
||||
rtc::scoped_refptr<VideoRtpTrackSource> source_;
|
||||
|
||||
@ -171,7 +171,7 @@ class AudioRtpSender : public DtmfProviderInterface,
|
||||
StatsCollector* stats_;
|
||||
rtc::scoped_refptr<AudioTrackInterface> track_;
|
||||
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
|
||||
rtc::Optional<std::string> last_transaction_id_;
|
||||
absl::optional<std::string> last_transaction_id_;
|
||||
uint32_t ssrc_ = 0;
|
||||
bool cached_track_enabled_ = false;
|
||||
bool stopped_ = false;
|
||||
@ -253,7 +253,7 @@ class VideoRtpSender : public ObserverInterface,
|
||||
std::vector<std::string> stream_ids_;
|
||||
cricket::VideoMediaChannel* media_channel_ = nullptr;
|
||||
rtc::scoped_refptr<VideoTrackInterface> track_;
|
||||
rtc::Optional<std::string> last_transaction_id_;
|
||||
absl::optional<std::string> last_transaction_id_;
|
||||
uint32_t ssrc_ = 0;
|
||||
VideoTrackInterface::ContentHint cached_track_content_hint_ =
|
||||
VideoTrackInterface::ContentHint::kNone;
|
||||
|
||||
@ -1101,7 +1101,7 @@ TEST_F(RtpSenderReceiverTest,
|
||||
video_track_->set_enabled(true);
|
||||
|
||||
// Sender is not ready to send (no SSRC) so no option should have been set.
|
||||
EXPECT_EQ(rtc::nullopt, video_media_channel_->options().is_screencast);
|
||||
EXPECT_EQ(absl::nullopt, video_media_channel_->options().is_screencast);
|
||||
|
||||
// Verify that the content hint is accounted for when video_rtp_sender_ does
|
||||
// get enabled.
|
||||
|
||||
@ -148,7 +148,7 @@ cricket::MediaType RtpTransceiver::media_type() const {
|
||||
return media_type_;
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> RtpTransceiver::mid() const {
|
||||
absl::optional<std::string> RtpTransceiver::mid() const {
|
||||
return mid_;
|
||||
}
|
||||
|
||||
@ -203,7 +203,7 @@ void RtpTransceiver::SetDirection(RtpTransceiverDirection new_direction) {
|
||||
SignalNegotiationNeeded();
|
||||
}
|
||||
|
||||
rtc::Optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
|
||||
absl::optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
|
||||
const {
|
||||
return current_direction_;
|
||||
}
|
||||
@ -216,7 +216,7 @@ void RtpTransceiver::Stop() {
|
||||
receiver->internal()->Stop();
|
||||
}
|
||||
stopped_ = true;
|
||||
current_direction_ = rtc::nullopt;
|
||||
current_direction_ = absl::nullopt;
|
||||
}
|
||||
|
||||
void RtpTransceiver::SetCodecPreferences(
|
||||
|
||||
@ -119,15 +119,15 @@ class RtpTransceiver final
|
||||
// when setting a local offer we need a way to remember which transceiver was
|
||||
// used to create which media section in the offer. Storing the mline index
|
||||
// in CreateOffer is specified in JSEP to allow us to do that.
|
||||
rtc::Optional<size_t> mline_index() const { return mline_index_; }
|
||||
void set_mline_index(rtc::Optional<size_t> mline_index) {
|
||||
absl::optional<size_t> mline_index() const { return mline_index_; }
|
||||
void set_mline_index(absl::optional<size_t> mline_index) {
|
||||
mline_index_ = mline_index;
|
||||
}
|
||||
|
||||
// Sets the MID for this transceiver. If the MID is not null, then the
|
||||
// transceiver is considered "associated" with the media section that has the
|
||||
// same MID.
|
||||
void set_mid(const rtc::Optional<std::string>& mid) { mid_ = mid; }
|
||||
void set_mid(const absl::optional<std::string>& mid) { mid_ = mid; }
|
||||
|
||||
// Sets the intended direction for this transceiver. Intended to be used
|
||||
// internally over SetDirection since this does not trigger a negotiation
|
||||
@ -160,13 +160,13 @@ class RtpTransceiver final
|
||||
|
||||
// RtpTransceiverInterface implementation.
|
||||
cricket::MediaType media_type() const override;
|
||||
rtc::Optional<std::string> mid() const override;
|
||||
absl::optional<std::string> mid() const override;
|
||||
rtc::scoped_refptr<RtpSenderInterface> sender() const override;
|
||||
rtc::scoped_refptr<RtpReceiverInterface> receiver() const override;
|
||||
bool stopped() const override;
|
||||
RtpTransceiverDirection direction() const override;
|
||||
void SetDirection(RtpTransceiverDirection new_direction) override;
|
||||
rtc::Optional<RtpTransceiverDirection> current_direction() const override;
|
||||
absl::optional<RtpTransceiverDirection> current_direction() const override;
|
||||
void Stop() override;
|
||||
void SetCodecPreferences(rtc::ArrayView<RtpCodecCapability> codecs) override;
|
||||
|
||||
@ -183,9 +183,9 @@ class RtpTransceiver final
|
||||
|
||||
bool stopped_ = false;
|
||||
RtpTransceiverDirection direction_ = RtpTransceiverDirection::kInactive;
|
||||
rtc::Optional<RtpTransceiverDirection> current_direction_;
|
||||
rtc::Optional<std::string> mid_;
|
||||
rtc::Optional<size_t> mline_index_;
|
||||
absl::optional<RtpTransceiverDirection> current_direction_;
|
||||
absl::optional<std::string> mid_;
|
||||
absl::optional<size_t> mline_index_;
|
||||
bool created_by_addtrack_ = false;
|
||||
bool has_ever_been_used_to_send_ = false;
|
||||
|
||||
@ -195,13 +195,13 @@ class RtpTransceiver final
|
||||
BEGIN_SIGNALING_PROXY_MAP(RtpTransceiver)
|
||||
PROXY_SIGNALING_THREAD_DESTRUCTOR()
|
||||
PROXY_CONSTMETHOD0(cricket::MediaType, media_type);
|
||||
PROXY_CONSTMETHOD0(rtc::Optional<std::string>, mid);
|
||||
PROXY_CONSTMETHOD0(absl::optional<std::string>, mid);
|
||||
PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender);
|
||||
PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver);
|
||||
PROXY_CONSTMETHOD0(bool, stopped);
|
||||
PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction);
|
||||
PROXY_METHOD1(void, SetDirection, RtpTransceiverDirection);
|
||||
PROXY_CONSTMETHOD0(rtc::Optional<RtpTransceiverDirection>, current_direction);
|
||||
PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction);
|
||||
PROXY_METHOD0(void, Stop);
|
||||
PROXY_METHOD1(void, SetCodecPreferences, rtc::ArrayView<RtpCodecCapability>);
|
||||
END_PROXY_MAP();
|
||||
|
||||
@ -39,7 +39,7 @@ void RtpTransport::SetRtpPacketTransport(
|
||||
rtp_packet_transport_->SignalWritableState.disconnect(this);
|
||||
rtp_packet_transport_->SignalSentPacket.disconnect(this);
|
||||
// Reset the network route of the old transport.
|
||||
SignalNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute>());
|
||||
SignalNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
|
||||
}
|
||||
if (new_packet_transport) {
|
||||
new_packet_transport->SignalReadyToSend.connect(
|
||||
@ -75,7 +75,7 @@ void RtpTransport::SetRtcpPacketTransport(
|
||||
rtcp_packet_transport_->SignalWritableState.disconnect(this);
|
||||
rtcp_packet_transport_->SignalSentPacket.disconnect(this);
|
||||
// Reset the network route of the old transport.
|
||||
SignalNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute>());
|
||||
SignalNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
|
||||
}
|
||||
if (new_packet_transport) {
|
||||
new_packet_transport->SignalReadyToSend.connect(
|
||||
@ -217,7 +217,7 @@ void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
|
||||
}
|
||||
|
||||
void RtpTransport::OnNetworkRouteChanged(
|
||||
rtc::Optional<rtc::NetworkRoute> network_route) {
|
||||
absl::optional<rtc::NetworkRoute> network_route) {
|
||||
SignalNetworkRouteChanged(network_route);
|
||||
}
|
||||
|
||||
|
||||
@ -100,7 +100,7 @@ class RtpTransport : public RtpTransportInternal {
|
||||
|
||||
// Overridden by SrtpTransport.
|
||||
virtual void OnNetworkRouteChanged(
|
||||
rtc::Optional<rtc::NetworkRoute> network_route);
|
||||
absl::optional<rtc::NetworkRoute> network_route);
|
||||
virtual void OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
|
||||
const rtc::PacketTime& packet_time);
|
||||
virtual void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
|
||||
|
||||
@ -78,8 +78,8 @@ class SignalObserver : public sigslot::has_slots<> {
|
||||
bool ready() const { return ready_; }
|
||||
void OnReadyToSend(bool ready) { ready_ = ready; }
|
||||
|
||||
rtc::Optional<rtc::NetworkRoute> network_route() { return network_route_; }
|
||||
void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route) {
|
||||
absl::optional<rtc::NetworkRoute> network_route() { return network_route_; }
|
||||
void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route) {
|
||||
network_route_ = std::move(network_route);
|
||||
}
|
||||
|
||||
@ -102,7 +102,7 @@ class SignalObserver : public sigslot::has_slots<> {
|
||||
int rtcp_transport_sent_count_ = 0;
|
||||
RtpTransport* transport_ = nullptr;
|
||||
bool ready_ = false;
|
||||
rtc::Optional<rtc::NetworkRoute> network_route_;
|
||||
absl::optional<rtc::NetworkRoute> network_route_;
|
||||
};
|
||||
|
||||
TEST(RtpTransportTest, SettingRtcpAndRtpSignalsReady) {
|
||||
@ -184,7 +184,7 @@ TEST(RtpTransportTest, SetRtpTransportWithNetworkRouteChanged) {
|
||||
network_route.remote_network_id = kRemoteNetId;
|
||||
network_route.last_sent_packet_id = kLastPacketId;
|
||||
network_route.packet_overhead = kTransportOverheadPerPacket;
|
||||
fake_rtp.SetNetworkRoute(rtc::Optional<rtc::NetworkRoute>(network_route));
|
||||
fake_rtp.SetNetworkRoute(absl::optional<rtc::NetworkRoute>(network_route));
|
||||
transport.SetRtpPacketTransport(&fake_rtp);
|
||||
ASSERT_TRUE(observer.network_route());
|
||||
EXPECT_EQ(network_route, *(observer.network_route()));
|
||||
@ -211,7 +211,7 @@ TEST(RtpTransportTest, SetRtcpTransportWithNetworkRouteChanged) {
|
||||
network_route.remote_network_id = kRemoteNetId;
|
||||
network_route.last_sent_packet_id = kLastPacketId;
|
||||
network_route.packet_overhead = kTransportOverheadPerPacket;
|
||||
fake_rtcp.SetNetworkRoute(rtc::Optional<rtc::NetworkRoute>(network_route));
|
||||
fake_rtcp.SetNetworkRoute(absl::optional<rtc::NetworkRoute>(network_route));
|
||||
transport.SetRtcpPacketTransport(&fake_rtcp);
|
||||
ASSERT_TRUE(observer.network_route());
|
||||
EXPECT_EQ(network_route, *(observer.network_route()));
|
||||
|
||||
@ -65,7 +65,7 @@ class RtpTransportInternal : public SrtpTransportInterface,
|
||||
|
||||
// Called whenever the network route of the P2P layer transport changes.
|
||||
// The argument is an optional network route.
|
||||
sigslot::signal1<rtc::Optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
|
||||
sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
|
||||
|
||||
// Called whenever a transport's writable state might change. The argument is
|
||||
// true if the transport is writable, otherwise it is false.
|
||||
|
||||
@ -186,8 +186,8 @@ bool SrtpFilter::ResetParams() {
|
||||
offer_params_.clear();
|
||||
applied_send_params_ = CryptoParams();
|
||||
applied_recv_params_ = CryptoParams();
|
||||
send_cipher_suite_ = rtc::nullopt;
|
||||
recv_cipher_suite_ = rtc::nullopt;
|
||||
send_cipher_suite_ = absl::nullopt;
|
||||
recv_cipher_suite_ = absl::nullopt;
|
||||
send_key_.Clear();
|
||||
recv_key_.Clear();
|
||||
state_ = ST_INIT;
|
||||
|
||||
@ -17,10 +17,10 @@
|
||||
#include <string>
|
||||
#include <vector>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/cryptoparams.h"
|
||||
#include "api/jsep.h"
|
||||
#include "api/optional.h"
|
||||
#include "pc/sessiondescription.h"
|
||||
#include "rtc_base/buffer.h"
|
||||
#include "rtc_base/constructormagic.h"
|
||||
@ -78,8 +78,8 @@ class SrtpFilter {
|
||||
|
||||
bool ResetParams();
|
||||
|
||||
rtc::Optional<int> send_cipher_suite() { return send_cipher_suite_; }
|
||||
rtc::Optional<int> recv_cipher_suite() { return recv_cipher_suite_; }
|
||||
absl::optional<int> send_cipher_suite() { return send_cipher_suite_; }
|
||||
absl::optional<int> recv_cipher_suite() { return recv_cipher_suite_; }
|
||||
|
||||
rtc::ArrayView<const uint8_t> send_key() { return send_key_; }
|
||||
rtc::ArrayView<const uint8_t> recv_key() { return recv_key_; }
|
||||
@ -135,8 +135,8 @@ class SrtpFilter {
|
||||
std::vector<CryptoParams> offer_params_;
|
||||
CryptoParams applied_send_params_;
|
||||
CryptoParams applied_recv_params_;
|
||||
rtc::Optional<int> send_cipher_suite_;
|
||||
rtc::Optional<int> recv_cipher_suite_;
|
||||
absl::optional<int> send_cipher_suite_;
|
||||
absl::optional<int> recv_cipher_suite_;
|
||||
rtc::ZeroOnFreeBuffer<uint8_t> send_key_;
|
||||
rtc::ZeroOnFreeBuffer<uint8_t> recv_key_;
|
||||
};
|
||||
|
||||
@ -237,7 +237,7 @@ void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
|
||||
}
|
||||
|
||||
void SrtpTransport::OnNetworkRouteChanged(
|
||||
rtc::Optional<rtc::NetworkRoute> network_route) {
|
||||
absl::optional<rtc::NetworkRoute> network_route) {
|
||||
// Only append the SRTP overhead when there is a selected network route.
|
||||
if (network_route) {
|
||||
int srtp_overhead = 0;
|
||||
|
||||
@ -117,7 +117,7 @@ class SrtpTransport : public RtpTransport {
|
||||
void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
|
||||
const rtc::PacketTime& packet_time) override;
|
||||
void OnNetworkRouteChanged(
|
||||
rtc::Optional<rtc::NetworkRoute> network_route) override;
|
||||
absl::optional<rtc::NetworkRoute> network_route) override;
|
||||
|
||||
// Override the RtpTransport::OnWritableState.
|
||||
void OnWritableState(rtc::PacketTransportInternal* packet_transport) override;
|
||||
@ -151,10 +151,10 @@ class SrtpTransport : public RtpTransport {
|
||||
std::unique_ptr<cricket::SrtpSession> send_rtcp_session_;
|
||||
std::unique_ptr<cricket::SrtpSession> recv_rtcp_session_;
|
||||
|
||||
rtc::Optional<cricket::CryptoParams> send_params_;
|
||||
rtc::Optional<cricket::CryptoParams> recv_params_;
|
||||
rtc::Optional<int> send_cipher_suite_;
|
||||
rtc::Optional<int> recv_cipher_suite_;
|
||||
absl::optional<cricket::CryptoParams> send_params_;
|
||||
absl::optional<cricket::CryptoParams> recv_params_;
|
||||
absl::optional<int> send_cipher_suite_;
|
||||
absl::optional<int> recv_cipher_suite_;
|
||||
rtc::ZeroOnFreeBuffer<uint8_t> send_key_;
|
||||
rtc::ZeroOnFreeBuffer<uint8_t> recv_key_;
|
||||
|
||||
|
||||
@ -266,12 +266,12 @@ class FakePeerConnectionBase : public PeerConnectionInternal {
|
||||
return {};
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> sctp_content_name() const override {
|
||||
return rtc::nullopt;
|
||||
absl::optional<std::string> sctp_content_name() const override {
|
||||
return absl::nullopt;
|
||||
}
|
||||
|
||||
rtc::Optional<std::string> sctp_transport_name() const override {
|
||||
return rtc::nullopt;
|
||||
absl::optional<std::string> sctp_transport_name() const override {
|
||||
return absl::nullopt;
|
||||
}
|
||||
|
||||
std::map<std::string, std::string> GetTransportNamesByMid() const override {
|
||||
|
||||
@ -45,7 +45,7 @@ class FakeVoiceMediaChannelForStats : public cricket::FakeVoiceMediaChannel {
|
||||
}
|
||||
|
||||
private:
|
||||
rtc::Optional<cricket::VoiceMediaInfo> stats_;
|
||||
absl::optional<cricket::VoiceMediaInfo> stats_;
|
||||
};
|
||||
|
||||
// Fake VideoMediaChannel where the result of GetStats can be configured.
|
||||
@ -68,7 +68,7 @@ class FakeVideoMediaChannelForStats : public cricket::FakeVideoMediaChannel {
|
||||
}
|
||||
|
||||
private:
|
||||
rtc::Optional<cricket::VideoMediaInfo> stats_;
|
||||
absl::optional<cricket::VideoMediaInfo> stats_;
|
||||
};
|
||||
|
||||
constexpr bool kDefaultRtcpMuxRequired = true;
|
||||
|
||||
@ -140,7 +140,7 @@ class FakeRTCCertificateGenerator
|
||||
|
||||
void GenerateCertificateAsync(
|
||||
const rtc::KeyParams& key_params,
|
||||
const rtc::Optional<uint64_t>& expires_ms,
|
||||
const absl::optional<uint64_t>& expires_ms,
|
||||
const rtc::scoped_refptr<rtc::RTCCertificateGeneratorCallback>& callback)
|
||||
override {
|
||||
// The certificates are created from constant PEM strings and use its coded
|
||||
|
||||
@ -40,8 +40,8 @@ class FakeSctpTransport : public cricket::SctpTransportInternal {
|
||||
int remote_port() const { return *remote_port_; }
|
||||
|
||||
private:
|
||||
rtc::Optional<int> local_port_;
|
||||
rtc::Optional<int> remote_port_;
|
||||
absl::optional<int> local_port_;
|
||||
absl::optional<int> remote_port_;
|
||||
};
|
||||
|
||||
class FakeSctpTransportFactory : public cricket::SctpTransportInternalFactory {
|
||||
|
||||
@ -293,7 +293,7 @@ class MockSetRemoteDescriptionObserver
|
||||
|
||||
private:
|
||||
// Set on complete, on success this is set to an RTCError::OK() error.
|
||||
rtc::Optional<RTCError> error_;
|
||||
absl::optional<RTCError> error_;
|
||||
};
|
||||
|
||||
class MockDataChannelObserver : public webrtc::DataChannelObserver {
|
||||
|
||||
@ -266,11 +266,11 @@ rtc::scoped_refptr<VideoTrackInterface> TrackMediaInfoMap::GetVideoTrack(
|
||||
return FindValueOrNull(video_track_by_receiver_info_, &video_receiver_info);
|
||||
}
|
||||
|
||||
rtc::Optional<int> TrackMediaInfoMap::GetAttachmentIdByTrack(
|
||||
absl::optional<int> TrackMediaInfoMap::GetAttachmentIdByTrack(
|
||||
const MediaStreamTrackInterface* track) const {
|
||||
auto it = attachment_id_by_track_.find(track);
|
||||
return it != attachment_id_by_track_.end() ? rtc::Optional<int>(it->second)
|
||||
: rtc::nullopt;
|
||||
return it != attachment_id_by_track_.end() ? absl::optional<int>(it->second)
|
||||
: absl::nullopt;
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
@ -79,12 +79,12 @@ class TrackMediaInfoMap {
|
||||
// It is not going to work if a track is attached multiple times, and
|
||||
// it is not going to work if a received track is attached as a sending
|
||||
// track (loopback).
|
||||
rtc::Optional<int> GetAttachmentIdByTrack(
|
||||
absl::optional<int> GetAttachmentIdByTrack(
|
||||
const MediaStreamTrackInterface* track) const;
|
||||
|
||||
private:
|
||||
rtc::Optional<std::string> voice_mid_;
|
||||
rtc::Optional<std::string> video_mid_;
|
||||
absl::optional<std::string> voice_mid_;
|
||||
absl::optional<std::string> video_mid_;
|
||||
std::unique_ptr<cricket::VoiceMediaInfo> voice_media_info_;
|
||||
std::unique_ptr<cricket::VideoMediaInfo> video_media_info_;
|
||||
// These maps map tracks (identified by a pointer) to their corresponding info
|
||||
|
||||
@ -408,7 +408,7 @@ TEST_F(TrackMediaInfoMapTest, GetAttachmentIdByTrack) {
|
||||
CreateMap();
|
||||
EXPECT_EQ(rtp_senders_[0]->AttachmentId(),
|
||||
map_->GetAttachmentIdByTrack(local_audio_track_));
|
||||
EXPECT_EQ(rtc::nullopt, map_->GetAttachmentIdByTrack(local_video_track_));
|
||||
EXPECT_EQ(absl::nullopt, map_->GetAttachmentIdByTrack(local_video_track_));
|
||||
}
|
||||
|
||||
// Death tests.
|
||||
|
||||
@ -242,7 +242,7 @@ const cricket::VideoFormat& GetBestCaptureFormat(
|
||||
// Return false if the key is mandatory, and the value is invalid.
|
||||
bool ExtractOption(const MediaConstraintsInterface* all_constraints,
|
||||
const std::string& key,
|
||||
rtc::Optional<bool>* option) {
|
||||
absl::optional<bool>* option) {
|
||||
size_t mandatory = 0;
|
||||
bool value;
|
||||
if (FindConstraint(all_constraints, key, &value, &mandatory)) {
|
||||
|
||||
@ -48,7 +48,9 @@ class VideoCapturerTrackSource : public VideoTrackSource,
|
||||
bool remote);
|
||||
|
||||
bool is_screencast() const final { return video_capturer_->IsScreencast(); }
|
||||
rtc::Optional<bool> needs_denoising() const final { return needs_denoising_; }
|
||||
absl::optional<bool> needs_denoising() const final {
|
||||
return needs_denoising_;
|
||||
}
|
||||
|
||||
bool GetStats(Stats* stats) final;
|
||||
|
||||
@ -76,7 +78,7 @@ class VideoCapturerTrackSource : public VideoTrackSource,
|
||||
std::unique_ptr<cricket::VideoCapturer> video_capturer_;
|
||||
bool started_;
|
||||
cricket::VideoFormat format_;
|
||||
rtc::Optional<bool> needs_denoising_;
|
||||
absl::optional<bool> needs_denoising_;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
@ -322,7 +322,7 @@ TEST_F(VideoCapturerTrackSourceTest, SetValidDenoisingConstraint) {
|
||||
TEST_F(VideoCapturerTrackSourceTest, NoiseReductionConstraintNotSet) {
|
||||
FakeConstraints constraints;
|
||||
CreateVideoCapturerSource(&constraints);
|
||||
EXPECT_EQ(rtc::nullopt, source_->needs_denoising());
|
||||
EXPECT_EQ(absl::nullopt, source_->needs_denoising());
|
||||
}
|
||||
|
||||
TEST_F(VideoCapturerTrackSourceTest,
|
||||
@ -357,7 +357,7 @@ TEST_F(VideoCapturerTrackSourceTest, NoiseReductionAndInvalidKeyMandatory) {
|
||||
|
||||
EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
|
||||
kMaxWaitMs);
|
||||
EXPECT_EQ(rtc::nullopt, source_->needs_denoising());
|
||||
EXPECT_EQ(absl::nullopt, source_->needs_denoising());
|
||||
}
|
||||
|
||||
TEST_F(VideoCapturerTrackSourceTest, InvalidDenoisingValueOptional) {
|
||||
@ -370,12 +370,13 @@ TEST_F(VideoCapturerTrackSourceTest, InvalidDenoisingValueOptional) {
|
||||
EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
|
||||
kMaxWaitMs);
|
||||
|
||||
EXPECT_EQ(rtc::nullopt, source_->needs_denoising());
|
||||
EXPECT_EQ(absl::nullopt, source_->needs_denoising());
|
||||
}
|
||||
|
||||
TEST_F(VideoCapturerTrackSourceTest, InvalidDenoisingValueMandatory) {
|
||||
FakeConstraints constraints;
|
||||
// Optional constraints should be ignored if the mandatory constraints fail.
|
||||
// absl::optional constraints should be ignored if the mandatory constraints
|
||||
// fail.
|
||||
constraints.AddOptional(MediaConstraintsInterface::kNoiseReduction, "false");
|
||||
// Values are case-sensitive and must be all lower-case.
|
||||
constraints.AddMandatory(MediaConstraintsInterface::kNoiseReduction, "True");
|
||||
@ -384,7 +385,7 @@ TEST_F(VideoCapturerTrackSourceTest, InvalidDenoisingValueMandatory) {
|
||||
|
||||
EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
|
||||
kMaxWaitMs);
|
||||
EXPECT_EQ(rtc::nullopt, source_->needs_denoising());
|
||||
EXPECT_EQ(absl::nullopt, source_->needs_denoising());
|
||||
}
|
||||
|
||||
TEST_F(VideoCapturerTrackSourceTest, MixedOptionsAndConstraints) {
|
||||
|
||||
@ -30,7 +30,9 @@ class VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
|
||||
bool remote() const override { return remote_; }
|
||||
|
||||
bool is_screencast() const override { return false; }
|
||||
rtc::Optional<bool> needs_denoising() const override { return rtc::nullopt; }
|
||||
absl::optional<bool> needs_denoising() const override {
|
||||
return absl::nullopt;
|
||||
}
|
||||
|
||||
bool GetStats(Stats* stats) override { return false; }
|
||||
|
||||
|
||||
@ -2032,7 +2032,7 @@ bool ParseSessionDescription(const std::string& message,
|
||||
std::string(), error);
|
||||
}
|
||||
|
||||
// Optional lines
|
||||
// absl::optional lines
|
||||
// Those are the optional lines, so shouldn't return false if not present.
|
||||
// RFC 4566
|
||||
// i=* (session information)
|
||||
|
||||
@ -175,7 +175,7 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
|
||||
|
||||
// Request certificate. This happens asynchronously, so that the caller gets
|
||||
// a chance to connect to |SignalCertificateReady|.
|
||||
cert_generator_->GenerateCertificateAsync(key_params, rtc::nullopt,
|
||||
cert_generator_->GenerateCertificateAsync(key_params, absl::nullopt,
|
||||
callback);
|
||||
}
|
||||
}
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user