diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc index a8659de5f9..6e7ca6d0b5 100644 --- a/pc/audio_rtp_receiver.cc +++ b/pc/audio_rtp_receiver.cc @@ -278,7 +278,7 @@ std::vector AudioRtpReceiver::GetSources() const { } void AudioRtpReceiver::SetDepacketizerToDecoderFrameTransformer( - rtc::scoped_refptr frame_transformer) { + rtc::scoped_refptr frame_transformer) { RTC_DCHECK_RUN_ON(worker_thread_); if (media_channel_) { media_channel_->SetDepacketizerToDecoderFrameTransformer( diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h index 86c42d532a..36cbdffc35 100644 --- a/pc/audio_rtp_receiver.h +++ b/pc/audio_rtp_receiver.h @@ -118,8 +118,7 @@ class AudioRtpReceiver : public ObserverInterface, std::vector GetSources() const override; int AttachmentId() const override { return attachment_id_; } void SetDepacketizerToDecoderFrameTransformer( - rtc::scoped_refptr frame_transformer) - override; + rtc::scoped_refptr frame_transformer) override; private: void RestartMediaChannel(absl::optional ssrc) diff --git a/pc/audio_rtp_receiver_unittest.cc b/pc/audio_rtp_receiver_unittest.cc index 9eb20c982f..e031f90359 100644 --- a/pc/audio_rtp_receiver_unittest.cc +++ b/pc/audio_rtp_receiver_unittest.cc @@ -98,7 +98,7 @@ TEST_F(AudioRtpReceiverTest, VolumesSetBeforeStartingAreRespected) { // thread when a media channel pointer is passed to the receiver via the // constructor. TEST(AudioRtpReceiver, OnChangedNotificationsAfterConstruction) { - webrtc::test::RunLoop loop; + test::RunLoop loop; auto* thread = rtc::Thread::Current(); // Points to loop's thread. cricket::MockVoiceMediaReceiveChannelInterface receive_channel; auto receiver = rtc::make_ref_counted( diff --git a/pc/audio_track.h b/pc/audio_track.h index ae326b304b..92c3141d8a 100644 --- a/pc/audio_track.h +++ b/pc/audio_track.h @@ -58,7 +58,7 @@ class AudioTrack : public MediaStreamTrack, private: const rtc::scoped_refptr audio_source_; - RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker signaling_thread_checker_; + RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_; }; } // namespace webrtc diff --git a/pc/connection_context.h b/pc/connection_context.h index 399e7c2b45..af5b7a9b5f 100644 --- a/pc/connection_context.h +++ b/pc/connection_context.h @@ -138,7 +138,7 @@ class ConnectionContext final RTC_GUARDED_BY(signaling_thread_); std::unique_ptr default_network_manager_ RTC_GUARDED_BY(signaling_thread_); - std::unique_ptr const call_factory_ + std::unique_ptr const call_factory_ RTC_GUARDED_BY(worker_thread()); std::unique_ptr default_socket_factory_ diff --git a/pc/data_channel_controller_unittest.cc b/pc/data_channel_controller_unittest.cc index 3b8adb6819..7d4e60467e 100644 --- a/pc/data_channel_controller_unittest.cc +++ b/pc/data_channel_controller_unittest.cc @@ -27,7 +27,7 @@ namespace { using ::testing::NiceMock; using ::testing::Return; -class MockDataChannelTransport : public webrtc::DataChannelTransportInterface { +class MockDataChannelTransport : public DataChannelTransportInterface { public: ~MockDataChannelTransport() override {} diff --git a/pc/data_channel_integrationtest.cc b/pc/data_channel_integrationtest.cc index faec76d03e..5a8004c72a 100644 --- a/pc/data_channel_integrationtest.cc +++ b/pc/data_channel_integrationtest.cc @@ -90,7 +90,7 @@ class FakeClockForTest : public rtc::ScopedFakeClock { // Some things use a time of "0" as a special value, so we need to start out // the fake clock at a nonzero time. // TODO(deadbeef): Fix this. - AdvanceTime(webrtc::TimeDelta::Seconds(1)); + AdvanceTime(TimeDelta::Seconds(1)); } // Explicit handle. @@ -422,7 +422,7 @@ TEST_P(DataChannelIntegrationTest, CalleeClosesSctpDataChannel) { TEST_P(DataChannelIntegrationTest, SctpDataChannelConfigSentToOtherSide) { ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::DataChannelInit init; + DataChannelInit init; init.id = 53; init.maxRetransmits = 52; caller()->CreateDataChannel("data-channel", &init); @@ -453,7 +453,7 @@ TEST_P(DataChannelIntegrationTest, StressTestUnorderedSctpDataChannel) { // Normal procedure, but with unordered data channel config. ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::DataChannelInit init; + DataChannelInit init; init.ordered = false; caller()->CreateDataChannel(&init); caller()->CreateAndSetAndSignalOffer(); @@ -515,7 +515,7 @@ TEST_P(DataChannelIntegrationTest, StressTestOpenCloseChannelNoDelay) { const size_t kIterations = 10; bool has_negotiated = false; - webrtc::DataChannelInit init; + DataChannelInit init; for (size_t repeats = 0; repeats < kIterations; ++repeats) { RTC_LOG(LS_INFO) << "Iteration " << (repeats + 1) << "/" << kIterations; @@ -592,7 +592,7 @@ TEST_P(DataChannelIntegrationTest, StressTestOpenCloseChannelWithDelay) { const size_t kIterations = 10; bool has_negotiated = false; - webrtc::DataChannelInit init; + DataChannelInit init; for (size_t repeats = 0; repeats < kIterations; ++repeats) { RTC_LOG(LS_INFO) << "Iteration " << (repeats + 1) << "/" << kIterations; diff --git a/pc/data_channel_unittest.cc b/pc/data_channel_unittest.cc index 9b84a1be61..a27a66c3de 100644 --- a/pc/data_channel_unittest.cc +++ b/pc/data_channel_unittest.cc @@ -81,8 +81,7 @@ class SctpDataChannelTest : public ::testing::Test { controller_(new FakeDataChannelController(&network_thread_)) { network_thread_.Start(); inner_channel_ = controller_->CreateDataChannel("test", init_); - channel_ = - webrtc::SctpDataChannel::CreateProxy(inner_channel_, signaling_safety_); + channel_ = SctpDataChannel::CreateProxy(inner_channel_, signaling_safety_); } ~SctpDataChannelTest() override { run_loop_.Flush(); @@ -510,7 +509,7 @@ TEST_F(SctpDataChannelTest, LateCreatedChannelTransitionToOpen) { SetChannelReady(); InternalDataChannelInit init; init.id = 1; - auto dc = webrtc::SctpDataChannel::CreateProxy( + auto dc = SctpDataChannel::CreateProxy( controller_->CreateDataChannel("test1", init), signaling_safety_); EXPECT_EQ(DataChannelInterface::kOpen, dc->state()); } @@ -524,7 +523,7 @@ TEST_F(SctpDataChannelTest, SendUnorderedAfterReceivesOpenAck) { init.ordered = false; rtc::scoped_refptr dc = controller_->CreateDataChannel("test1", init); - auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_); + auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_); EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000); @@ -553,7 +552,7 @@ TEST_F(SctpDataChannelTest, DeprecatedSendUnorderedAfterReceivesOpenAck) { init.ordered = false; rtc::scoped_refptr dc = controller_->CreateDataChannel("test1", init); - auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_); + auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_); EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000); @@ -582,7 +581,7 @@ TEST_F(SctpDataChannelTest, SendUnorderedAfterReceiveData) { init.ordered = false; rtc::scoped_refptr dc = controller_->CreateDataChannel("test1", init); - auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_); + auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_); EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000); @@ -605,7 +604,7 @@ TEST_F(SctpDataChannelTest, DeprecatedSendUnorderedAfterReceiveData) { init.ordered = false; rtc::scoped_refptr dc = controller_->CreateDataChannel("test1", init); - auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_); + auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_); EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000); @@ -714,7 +713,7 @@ TEST_F(SctpDataChannelTest, NoMsgSentIfNegotiatedAndNotFromOpenMsg) { SetChannelReady(); rtc::scoped_refptr dc = controller_->CreateDataChannel("test1", config); - auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_); + auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_); EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000); EXPECT_EQ(0, controller_->last_sid()); @@ -779,7 +778,7 @@ TEST_F(SctpDataChannelTest, OpenAckSentIfCreatedFromOpenMessage) { SetChannelReady(); rtc::scoped_refptr dc = controller_->CreateDataChannel("test1", config); - auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_); + auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_); EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000); diff --git a/pc/ice_server_parsing_unittest.cc b/pc/ice_server_parsing_unittest.cc index 4356b1efb0..a38638e507 100644 --- a/pc/ice_server_parsing_unittest.cc +++ b/pc/ice_server_parsing_unittest.cc @@ -62,9 +62,7 @@ class IceServerParsingTest : public ::testing::Test { server.tls_cert_policy = tls_certificate_policy; server.hostname = hostname; servers.push_back(server); - return webrtc::ParseIceServersOrError(servers, &stun_servers_, - &turn_servers_) - .ok(); + return ParseIceServersOrError(servers, &stun_servers_, &turn_servers_).ok(); } protected: @@ -233,8 +231,7 @@ TEST_F(IceServerParsingTest, ParseMultipleUrls) { server.password = "bar"; servers.push_back(server); EXPECT_TRUE( - webrtc::ParseIceServersOrError(servers, &stun_servers_, &turn_servers_) - .ok()); + ParseIceServersOrError(servers, &stun_servers_, &turn_servers_).ok()); EXPECT_EQ(1U, stun_servers_.size()); EXPECT_EQ(1U, turn_servers_.size()); } diff --git a/pc/ice_transport_unittest.cc b/pc/ice_transport_unittest.cc index aaf9f2e57a..a42c107072 100644 --- a/pc/ice_transport_unittest.cc +++ b/pc/ice_transport_unittest.cc @@ -32,7 +32,7 @@ class IceTransportTest : public ::testing::Test { rtc::SocketServer* socket_server() const { return socket_server_.get(); } - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; private: std::unique_ptr socket_server_; diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc index 7c669a5ae3..2a701cce7f 100644 --- a/pc/jsep_transport_controller.cc +++ b/pc/jsep_transport_controller.cc @@ -148,7 +148,7 @@ JsepTransportController::GetRtcpDtlsTransport(const std::string& mid) const { return jsep_transport->rtcp_dtls_transport(); } -rtc::scoped_refptr +rtc::scoped_refptr JsepTransportController::LookupDtlsTransportByMid(const std::string& mid) { RTC_DCHECK_RUN_ON(network_thread_); auto jsep_transport = GetJsepTransportForMid(mid); @@ -383,7 +383,7 @@ RTCError JsepTransportController::RollbackTransports() { return RTCError::OK(); } -rtc::scoped_refptr +rtc::scoped_refptr JsepTransportController::CreateIceTransport(const std::string& transport_name, bool rtcp) { int component = rtcp ? cricket::ICE_CANDIDATE_COMPONENT_RTCP @@ -455,7 +455,7 @@ JsepTransportController::CreateDtlsTransport( return dtls; } -std::unique_ptr +std::unique_ptr JsepTransportController::CreateUnencryptedRtpTransport( const std::string& transport_name, rtc::PacketTransportInternal* rtp_packet_transport, @@ -470,13 +470,12 @@ JsepTransportController::CreateUnencryptedRtpTransport( return unencrypted_rtp_transport; } -std::unique_ptr -JsepTransportController::CreateSdesTransport( +std::unique_ptr JsepTransportController::CreateSdesTransport( const std::string& transport_name, cricket::DtlsTransportInternal* rtp_dtls_transport, cricket::DtlsTransportInternal* rtcp_dtls_transport) { RTC_DCHECK_RUN_ON(network_thread_); - auto srtp_transport = std::make_unique( + auto srtp_transport = std::make_unique( rtcp_dtls_transport == nullptr, *config_.field_trials); RTC_DCHECK(rtp_dtls_transport); srtp_transport->SetRtpPacketTransport(rtp_dtls_transport); @@ -489,13 +488,13 @@ JsepTransportController::CreateSdesTransport( return srtp_transport; } -std::unique_ptr +std::unique_ptr JsepTransportController::CreateDtlsSrtpTransport( const std::string& transport_name, cricket::DtlsTransportInternal* rtp_dtls_transport, cricket::DtlsTransportInternal* rtcp_dtls_transport) { RTC_DCHECK_RUN_ON(network_thread_); - auto dtls_srtp_transport = std::make_unique( + auto dtls_srtp_transport = std::make_unique( rtcp_dtls_transport == nullptr, *config_.field_trials); if (config_.enable_external_auth) { dtls_srtp_transport->EnableExternalAuth(); @@ -985,13 +984,12 @@ int JsepTransportController::GetRtpAbsSendTimeHeaderExtensionId( const cricket::MediaContentDescription* content_desc = content_info.media_description(); - const webrtc::RtpExtension* send_time_extension = - webrtc::RtpExtension::FindHeaderExtensionByUri( - content_desc->rtp_header_extensions(), - webrtc::RtpExtension::kAbsSendTimeUri, + const RtpExtension* send_time_extension = + RtpExtension::FindHeaderExtensionByUri( + content_desc->rtp_header_extensions(), RtpExtension::kAbsSendTimeUri, config_.crypto_options.srtp.enable_encrypted_rtp_header_extensions - ? webrtc::RtpExtension::kPreferEncryptedExtension - : webrtc::RtpExtension::kDiscardEncryptedExtension); + ? RtpExtension::kPreferEncryptedExtension + : RtpExtension::kDiscardEncryptedExtension); return send_time_extension ? send_time_extension->id : -1; } @@ -1039,7 +1037,7 @@ RTCError JsepTransportController::MaybeCreateJsepTransport( "SDES and DTLS-SRTP cannot be enabled at the same time."); } - rtc::scoped_refptr ice = + rtc::scoped_refptr ice = CreateIceTransport(content_info.name, /*rtcp=*/false); std::unique_ptr rtp_dtls_transport = @@ -1050,7 +1048,7 @@ RTCError JsepTransportController::MaybeCreateJsepTransport( std::unique_ptr sdes_transport; std::unique_ptr dtls_srtp_transport; - rtc::scoped_refptr rtcp_ice; + rtc::scoped_refptr rtcp_ice; if (config_.rtcp_mux_policy != PeerConnectionInterface::kRtcpMuxPolicyRequire && content_info.type == cricket::MediaProtocolType::kRtp) { @@ -1096,7 +1094,7 @@ RTCError JsepTransportController::MaybeCreateJsepTransport( OnRtcpPacketReceived_n(buffer, packet_time_ms); }); jsep_transport->rtp_transport()->SetUnDemuxableRtpPacketReceivedHandler( - [this](webrtc::RtpPacketReceived& packet) { + [this](RtpPacketReceived& packet) { RTC_DCHECK_RUN_ON(network_thread_); OnUnDemuxableRtpPacketReceived_n(packet); }); @@ -1421,7 +1419,7 @@ void JsepTransportController::OnRtcpPacketReceived_n( } void JsepTransportController::OnUnDemuxableRtpPacketReceived_n( - const webrtc::RtpPacketReceived& packet) { + const RtpPacketReceived& packet) { RTC_DCHECK(config_.un_demuxable_packet_handler); config_.un_demuxable_packet_handler(packet); } diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h index 5880e346cd..8f9b9c8491 100644 --- a/pc/jsep_transport_controller.h +++ b/pc/jsep_transport_controller.h @@ -112,7 +112,7 @@ class JsepTransportController : public sigslot::has_slots<> { rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; // `crypto_options` is used to determine if created DTLS transports // negotiate GCM crypto suites or not. - webrtc::CryptoOptions crypto_options; + CryptoOptions crypto_options; PeerConnectionInterface::BundlePolicy bundle_policy = PeerConnectionInterface::kBundlePolicyBalanced; PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy = @@ -120,7 +120,7 @@ class JsepTransportController : public sigslot::has_slots<> { bool disable_encryption = false; bool enable_external_auth = false; // Used to inject the ICE/DTLS transports created externally. - webrtc::IceTransportFactory* ice_transport_factory = nullptr; + IceTransportFactory* ice_transport_factory = nullptr; cricket::DtlsTransportFactory* dtls_transport_factory = nullptr; Observer* transport_observer = nullptr; // Must be provided and valid for the lifetime of the @@ -140,7 +140,7 @@ class JsepTransportController : public sigslot::has_slots<> { std::function on_dtls_handshake_error_; // Field trials. - const webrtc::FieldTrialsView* field_trials; + const FieldTrialsView* field_trials; }; // The ICE related events are fired on the `network_thread`. @@ -174,7 +174,7 @@ class JsepTransportController : public sigslot::has_slots<> { const cricket::DtlsTransportInternal* GetRtcpDtlsTransport( const std::string& mid) const; // Gets the externally sharable version of the DtlsTransport. - rtc::scoped_refptr LookupDtlsTransportByMid( + rtc::scoped_refptr LookupDtlsTransportByMid( const std::string& mid); rtc::scoped_refptr GetSctpTransport( const std::string& mid) const; @@ -399,19 +399,19 @@ class JsepTransportController : public sigslot::has_slots<> { std::unique_ptr CreateDtlsTransport( const cricket::ContentInfo& content_info, cricket::IceTransportInternal* ice); - rtc::scoped_refptr CreateIceTransport( + rtc::scoped_refptr CreateIceTransport( const std::string& transport_name, bool rtcp); - std::unique_ptr CreateUnencryptedRtpTransport( + std::unique_ptr CreateUnencryptedRtpTransport( const std::string& transport_name, rtc::PacketTransportInternal* rtp_packet_transport, rtc::PacketTransportInternal* rtcp_packet_transport); - std::unique_ptr CreateSdesTransport( + std::unique_ptr CreateSdesTransport( const std::string& transport_name, cricket::DtlsTransportInternal* rtp_dtls_transport, cricket::DtlsTransportInternal* rtcp_dtls_transport); - std::unique_ptr CreateDtlsSrtpTransport( + std::unique_ptr CreateDtlsSrtpTransport( const std::string& transport_name, cricket::DtlsTransportInternal* rtp_dtls_transport, cricket::DtlsTransportInternal* rtcp_dtls_transport); @@ -453,7 +453,7 @@ class JsepTransportController : public sigslot::has_slots<> { void OnRtcpPacketReceived_n(rtc::CopyOnWriteBuffer* packet, int64_t packet_time_us) RTC_RUN_ON(network_thread_); - void OnUnDemuxableRtpPacketReceived_n(const webrtc::RtpPacketReceived& packet) + void OnUnDemuxableRtpPacketReceived_n(const RtpPacketReceived& packet) RTC_RUN_ON(network_thread_); void OnDtlsHandshakeError(rtc::SSLHandshakeError error); diff --git a/pc/jsep_transport_controller_unittest.cc b/pc/jsep_transport_controller_unittest.cc index faa8842e35..f5e258c664 100644 --- a/pc/jsep_transport_controller_unittest.cc +++ b/pc/jsep_transport_controller_unittest.cc @@ -56,7 +56,7 @@ static const char kDataMid1[] = "data1"; namespace webrtc { -class FakeIceTransportFactory : public webrtc::IceTransportFactory { +class FakeIceTransportFactory : public IceTransportFactory { public: ~FakeIceTransportFactory() override = default; rtc::scoped_refptr CreateIceTransport( @@ -72,7 +72,7 @@ class FakeDtlsTransportFactory : public cricket::DtlsTransportFactory { public: std::unique_ptr CreateDtlsTransport( cricket::IceTransportInternal* ice, - const webrtc::CryptoOptions& crypto_options, + const CryptoOptions& crypto_options, rtc::SSLProtocolVersion max_version) override { return std::make_unique( static_cast(ice)); @@ -379,7 +379,7 @@ class JsepTransportControllerTest : public JsepTransportController::Observer, // Transport controller needs to be destroyed first, because it may issue // callbacks that modify the changed_*_by_mid in the destructor. std::unique_ptr transport_controller_; - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; }; TEST_F(JsepTransportControllerTest, GetRtpTransport) { @@ -425,7 +425,7 @@ TEST_F(JsepTransportControllerTest, GetDtlsTransport) { // and verify that the resulting container is empty. auto dtls_transport = transport_controller_->LookupDtlsTransportByMid(kVideoMid1); - webrtc::DtlsTransport* my_transport = + DtlsTransport* my_transport = static_cast(dtls_transport.get()); EXPECT_NE(nullptr, my_transport->internal()); transport_controller_.reset(); @@ -899,7 +899,7 @@ TEST_F(JsepTransportControllerTest, transport_controller_->GetDtlsTransport(kAudioMid1)); fake_audio_dtls->fake_ice_transport()->MaybeStartGathering(); fake_audio_dtls->fake_ice_transport()->SetTransportState( - webrtc::IceTransportState::kChecking, + IceTransportState::kChecking, cricket::IceTransportState::STATE_CONNECTING); EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking, ice_connection_state_, kTimeout); diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc index 3bc65ee3ee..98b7cb9677 100644 --- a/pc/legacy_stats_collector.cc +++ b/pc/legacy_stats_collector.cc @@ -355,9 +355,8 @@ void ExtractStats(const cricket::VideoReceiverInfo& info, report->AddInt64(StatsReport::kStatsValueNameInterframeDelayMaxMs, info.interframe_delay_max_ms); - report->AddString( - StatsReport::kStatsValueNameContentType, - webrtc::videocontenttypehelpers::ToString(info.content_type)); + report->AddString(StatsReport::kStatsValueNameContentType, + videocontenttypehelpers::ToString(info.content_type)); } void ExtractStats(const cricket::VideoSenderInfo& info, @@ -398,9 +397,8 @@ void ExtractStats(const cricket::VideoSenderInfo& info, for (const auto& i : ints) report->AddInt(i.name, i.value); report->AddString(StatsReport::kStatsValueNameMediaType, "video"); - report->AddString( - StatsReport::kStatsValueNameContentType, - webrtc::videocontenttypehelpers::ToString(info.content_type)); + report->AddString(StatsReport::kStatsValueNameContentType, + videocontenttypehelpers::ToString(info.content_type)); } void ExtractStats(const cricket::BandwidthEstimationInfo& info, @@ -1033,7 +1031,7 @@ void LegacyStatsCollector::ExtractBweInfo() { if (pc_->signaling_state() == PeerConnectionInterface::kClosed) return; - webrtc::Call::Stats call_stats = pc_->GetCallStats(); + Call::Stats call_stats = pc_->GetCallStats(); cricket::BandwidthEstimationInfo bwe_info; bwe_info.available_send_bandwidth = call_stats.send_bandwidth_bps; bwe_info.available_recv_bandwidth = call_stats.recv_bandwidth_bps; diff --git a/pc/legacy_stats_collector.h b/pc/legacy_stats_collector.h index e905b39d48..1c7aad0636 100644 --- a/pc/legacy_stats_collector.h +++ b/pc/legacy_stats_collector.h @@ -177,9 +177,9 @@ class LegacyStatsCollector : public LegacyStatsCollectorInterface { void ExtractMediaInfo( const std::map& transport_names_by_mid); void ExtractSenderInfo(); - webrtc::StatsReport* GetReport(const StatsReport::StatsType& type, - const std::string& id, - StatsReport::Direction direction); + StatsReport* GetReport(const StatsReport::StatsType& type, + const std::string& id, + StatsReport::Direction direction); // Helper method to get stats from the local audio tracks. void UpdateStatsFromExistingLocalAudioTracks(bool has_remote_tracks); diff --git a/pc/media_stream_unittest.cc b/pc/media_stream_unittest.cc index f55ea203fb..d6c79efae9 100644 --- a/pc/media_stream_unittest.cc +++ b/pc/media_stream_unittest.cc @@ -91,7 +91,7 @@ TEST_F(MediaStreamTest, GetTrackInfo) { ASSERT_EQ(1u, stream_->GetAudioTracks().size()); // Verify the video track. - scoped_refptr video_track( + scoped_refptr video_track( stream_->GetVideoTracks()[0]); EXPECT_EQ(0, video_track->id().compare(kVideoTrackId)); EXPECT_TRUE(video_track->enabled()); @@ -105,7 +105,7 @@ TEST_F(MediaStreamTest, GetTrackInfo) { EXPECT_TRUE(video_track->enabled()); // Verify the audio track. - scoped_refptr audio_track( + scoped_refptr audio_track( stream_->GetAudioTracks()[0]); EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId)); EXPECT_TRUE(audio_track->enabled()); @@ -139,14 +139,12 @@ TEST_F(MediaStreamTest, RemoveTrack) { } TEST_F(MediaStreamTest, ChangeVideoTrack) { - scoped_refptr video_track( - stream_->GetVideoTracks()[0]); + scoped_refptr video_track(stream_->GetVideoTracks()[0]); ChangeTrack(video_track.get()); } TEST_F(MediaStreamTest, ChangeAudioTrack) { - scoped_refptr audio_track( - stream_->GetAudioTracks()[0]); + scoped_refptr audio_track(stream_->GetAudioTracks()[0]); ChangeTrack(audio_track.get()); } diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc index 183cbeb7cd..46c28bba37 100644 --- a/pc/peer_connection.cc +++ b/pc/peer_connection.cc @@ -185,7 +185,7 @@ IceCandidatePairType GetIceCandidatePairCounter( absl::optional RTCConfigurationToIceConfigOptionalInt( int rtc_configuration_parameter) { if (rtc_configuration_parameter == - webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) { + PeerConnectionInterface::RTCConfiguration::kUndefined) { return absl::nullopt; } return rtc_configuration_parameter; @@ -449,7 +449,7 @@ bool PeerConnectionInterface::RTCConfiguration::operator==( absl::optional ice_unwritable_min_checks; absl::optional ice_inactive_timeout; absl::optional stun_candidate_keepalive_interval; - webrtc::TurnCustomizer* turn_customizer; + TurnCustomizer* turn_customizer; SdpSemantics sdp_semantics; absl::optional network_preference; bool active_reset_srtp_params; @@ -459,7 +459,7 @@ bool PeerConnectionInterface::RTCConfiguration::operator==( bool enable_implicit_rollback; absl::optional report_usage_pattern_delay_ms; absl::optional stable_writable_connection_ping_interval_ms; - webrtc::VpnPreference vpn_preference; + VpnPreference vpn_preference; std::vector vpn_list; PortAllocatorConfig port_allocator_config; absl::optional pacer_burst_interval; @@ -1685,7 +1685,7 @@ void PeerConnection::AddIceCandidate( std::function callback) { RTC_DCHECK_RUN_ON(signaling_thread()); sdp_handler_->AddIceCandidate(std::move(candidate), - [this, callback](webrtc::RTCError result) { + [this, callback](RTCError result) { ClearStatsCache(); callback(result); }); @@ -1789,7 +1789,7 @@ bool PeerConnection::StartRtcEventLog( std::unique_ptr output) { int64_t output_period_ms = 5000; if (trials().IsDisabled("WebRTC-RtcEventLogNewFormat")) { - output_period_ms = webrtc::RtcEventLog::kImmediateOutput; + output_period_ms = RtcEventLog::kImmediateOutput; } return StartRtcEventLog(std::move(output), output_period_ms); } @@ -2222,7 +2222,7 @@ bool PeerConnection::ReconfigurePortAllocator_n( IceTransportsType type, int candidate_pool_size, PortPrunePolicy turn_port_prune_policy, - webrtc::TurnCustomizer* turn_customizer, + TurnCustomizer* turn_customizer, absl::optional stun_candidate_keepalive_interval, bool have_local_description) { RTC_DCHECK_RUN_ON(network_thread()); diff --git a/pc/peer_connection.h b/pc/peer_connection.h index ea1a9d9d90..a345089191 100644 --- a/pc/peer_connection.h +++ b/pc/peer_connection.h @@ -163,7 +163,7 @@ class PeerConnection : public PeerConnectionInternal, const DataChannelInit* config) override; // WARNING: LEGACY. See peerconnectioninterface.h bool GetStats(StatsObserver* observer, - webrtc::MediaStreamTrackInterface* track, + MediaStreamTrackInterface* track, StatsOutputLevel level) override; // Spec-complaint GetStats(). See peerconnectioninterface.h void GetStats(RTCStatsCollectorCallback* callback) override; @@ -510,7 +510,7 @@ class PeerConnection : public PeerConnectionInternal, IceTransportsType type, int candidate_pool_size, PortPrunePolicy turn_port_prune_policy, - webrtc::TurnCustomizer* turn_customizer, + TurnCustomizer* turn_customizer, absl::optional stun_candidate_keepalive_interval, bool have_local_description); @@ -602,7 +602,7 @@ class PeerConnection : public PeerConnectionInternal, // a) Specified in PeerConnectionDependencies (owned). // b) Accessed via ConnectionContext (e.g PeerConnectionFactoryDependencies> // c) Created as Default (FieldTrialBasedConfig). - const webrtc::AlwaysValidPointer + const AlwaysValidPointer trials_; const PeerConnectionFactoryInterface::Options options_; PeerConnectionObserver* observer_ RTC_GUARDED_BY(signaling_thread()) = @@ -634,7 +634,7 @@ class PeerConnection : public PeerConnectionInternal, std::unique_ptr port_allocator_; // TODO(bugs.webrtc.org/9987): Accessed on both // signaling and network thread. - const std::unique_ptr + const std::unique_ptr ice_transport_factory_; // TODO(bugs.webrtc.org/9987): Accessed on the // signaling thread but the underlying raw // pointer is given to diff --git a/pc/peer_connection_crypto_unittest.cc b/pc/peer_connection_crypto_unittest.cc index dc350b2be0..a65988ab05 100644 --- a/pc/peer_connection_crypto_unittest.cc +++ b/pc/peer_connection_crypto_unittest.cc @@ -162,7 +162,7 @@ class PeerConnectionCryptoBaseTest : public ::testing::Test { return transport_info->description.connection_role; } - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; std::unique_ptr vss_; rtc::AutoSocketServerThread main_; rtc::scoped_refptr pc_factory_; diff --git a/pc/peer_connection_encodings_integrationtest.cc b/pc/peer_connection_encodings_integrationtest.cc index c7181c53ae..ae238671c2 100644 --- a/pc/peer_connection_encodings_integrationtest.cc +++ b/pc/peer_connection_encodings_integrationtest.cc @@ -77,18 +77,17 @@ struct StringParamToString { // RTX, RED and FEC are reliability mechanisms used in combinations with other // codecs, but are not themselves a specific codec. Typically you don't want to // filter these out of the list of codec preferences. -bool IsReliabilityMechanism(const webrtc::RtpCodecCapability& codec) { +bool IsReliabilityMechanism(const RtpCodecCapability& codec) { return absl::EqualsIgnoreCase(codec.name, cricket::kRtxCodecName) || absl::EqualsIgnoreCase(codec.name, cricket::kRedCodecName) || absl::EqualsIgnoreCase(codec.name, cricket::kUlpfecCodecName); } std::string GetCurrentCodecMimeType( - rtc::scoped_refptr report, - const webrtc::RTCOutboundRtpStreamStats& outbound_rtp) { + rtc::scoped_refptr report, + const RTCOutboundRtpStreamStats& outbound_rtp) { return outbound_rtp.codec_id.is_defined() - ? *report->GetAs(*outbound_rtp.codec_id) - ->mime_type + ? *report->GetAs(*outbound_rtp.codec_id)->mime_type : ""; } @@ -98,8 +97,8 @@ struct RidAndResolution { uint32_t height; }; -const webrtc::RTCOutboundRtpStreamStats* FindOutboundRtpByRid( - const std::vector& outbound_rtps, +const RTCOutboundRtpStreamStats* FindOutboundRtpByRid( + const std::vector& outbound_rtps, const absl::string_view& rid) { for (const auto* outbound_rtp : outbound_rtps) { if (outbound_rtp->rid.is_defined() && *outbound_rtp->rid == rid) { @@ -121,8 +120,8 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test { rtc::scoped_refptr CreatePc() { auto pc_wrapper = rtc::make_ref_counted( "pc", &pss_, background_thread_.get(), background_thread_.get()); - pc_wrapper->CreatePc({}, webrtc::CreateBuiltinAudioEncoderFactory(), - webrtc::CreateBuiltinAudioDecoderFactory()); + pc_wrapper->CreatePc({}, CreateBuiltinAudioEncoderFactory(), + CreateBuiltinAudioDecoderFactory()); return pc_wrapper; } @@ -130,10 +129,9 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test { rtc::scoped_refptr local, rtc::scoped_refptr remote, std::vector init_layers) { - rtc::scoped_refptr stream = - local->GetUserMedia( - /*audio=*/false, cricket::AudioOptions(), /*video=*/true, - {.width = 1280, .height = 720}); + rtc::scoped_refptr stream = local->GetUserMedia( + /*audio=*/false, cricket::AudioOptions(), /*video=*/true, + {.width = 1280, .height = 720}); rtc::scoped_refptr track = stream->GetVideoTracks()[0]; RTCErrorOr> @@ -973,8 +971,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO); rtc::scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); EXPECT_FALSE(parameters.encodings[0].codec.has_value()); } @@ -986,8 +983,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO); rtc::scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); EXPECT_FALSE(parameters.encodings[0].codec.has_value()); } @@ -997,19 +993,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr stream = + rtc::scoped_refptr stream = local_pc_wrapper->GetUserMedia( /*audio=*/true, {}, /*video=*/false, {}); rtc::scoped_refptr track = stream->GetAudioTracks()[0]; - absl::optional pcmu = + absl::optional pcmu = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "pcmu"); ASSERT_TRUE(pcmu); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = pcmu; init.send_encodings.push_back(encoding_parameters); @@ -1017,8 +1013,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->pc()->AddTransceiver(track, init); rtc::scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); EXPECT_EQ(*parameters.encodings[0].codec, *pcmu); NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); @@ -1039,19 +1034,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr stream = + rtc::scoped_refptr stream = local_pc_wrapper->GetUserMedia( /*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720}); rtc::scoped_refptr track = stream->GetVideoTracks()[0]; - absl::optional vp9 = + absl::optional vp9 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp9"); ASSERT_TRUE(vp9); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = vp9; encoding_parameters.scalability_mode = "L3T3"; init.send_encodings.push_back(encoding_parameters); @@ -1060,8 +1055,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->pc()->AddTransceiver(track, init); rtc::scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); EXPECT_EQ(*parameters.encodings[0].codec, *vp9); NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); @@ -1087,20 +1081,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr stream = + rtc::scoped_refptr stream = local_pc_wrapper->GetUserMedia( /*audio=*/true, {}, /*video=*/false, {}); rtc::scoped_refptr track = stream->GetAudioTracks()[0]; - absl::optional pcmu = + absl::optional pcmu = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "pcmu"); auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); rtc::scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = pcmu; EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok()); @@ -1125,12 +1118,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr stream = + rtc::scoped_refptr stream = local_pc_wrapper->GetUserMedia( /*audio=*/true, {}, /*video=*/false, {}); rtc::scoped_refptr track = stream->GetAudioTracks()[0]; - absl::optional pcmu = + absl::optional pcmu = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "pcmu"); @@ -1150,8 +1143,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, EXPECT_STRCASENE(("audio/" + pcmu->name).c_str(), codec_name.c_str()); std::string last_codec_id = outbound_rtps[0]->codec_id.value(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = pcmu; EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok()); @@ -1174,20 +1166,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr stream = + rtc::scoped_refptr stream = local_pc_wrapper->GetUserMedia( /*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720}); rtc::scoped_refptr track = stream->GetVideoTracks()[0]; - absl::optional vp9 = + absl::optional vp9 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp9"); auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); rtc::scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = vp9; parameters.encodings[0].scalability_mode = "L3T3"; EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok()); @@ -1218,12 +1209,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr stream = + rtc::scoped_refptr stream = local_pc_wrapper->GetUserMedia( /*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720}); rtc::scoped_refptr track = stream->GetVideoTracks()[0]; - absl::optional vp9 = + absl::optional vp9 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp9"); @@ -1243,8 +1234,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, EXPECT_STRCASENE(("audio/" + vp9->name).c_str(), codec_name.c_str()); std::string last_codec_id = outbound_rtps[0]->codec_id.value(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = vp9; parameters.encodings[0].scalability_mode = "L3T3"; EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok()); @@ -1269,15 +1259,15 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, AddTransceiverRejectsUnknownCodecParameterAudio) { rtc::scoped_refptr local_pc_wrapper = CreatePc(); - webrtc::RtpCodec dummy_codec; + RtpCodec dummy_codec; dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; dummy_codec.num_channels = 2; - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = dummy_codec; init.send_encodings.push_back(encoding_parameters); @@ -1292,14 +1282,14 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, AddTransceiverRejectsUnknownCodecParameterVideo) { rtc::scoped_refptr local_pc_wrapper = CreatePc(); - webrtc::RtpCodec dummy_codec; + RtpCodec dummy_codec; dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = dummy_codec; init.send_encodings.push_back(encoding_parameters); @@ -1314,7 +1304,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsUnknownCodecParameterAudio) { rtc::scoped_refptr local_pc_wrapper = CreatePc(); - webrtc::RtpCodec dummy_codec; + RtpCodec dummy_codec; dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; @@ -1326,8 +1316,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = dummy_codec; RTCError error = audio_transceiver->sender()->SetParameters(parameters); EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); @@ -1337,7 +1326,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsUnknownCodecParameterVideo) { rtc::scoped_refptr local_pc_wrapper = CreatePc(); - webrtc::RtpCodec dummy_codec; + RtpCodec dummy_codec; dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; @@ -1348,8 +1337,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = dummy_codec; RTCError error = video_transceiver->sender()->SetParameters(parameters); EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); @@ -1359,12 +1347,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsNonPreferredCodecParameterAudio) { rtc::scoped_refptr local_pc_wrapper = CreatePc(); - absl::optional opus = + absl::optional opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - std::vector not_opus_codecs = + std::vector not_opus_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; @@ -1382,8 +1370,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, transceiver_or_error.MoveValue(); ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok()); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = opus; RTCError error = audio_transceiver->sender()->SetParameters(parameters); EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); @@ -1393,12 +1380,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsNonPreferredCodecParameterVideo) { rtc::scoped_refptr local_pc_wrapper = CreatePc(); - absl::optional vp8 = + absl::optional vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - std::vector not_vp8_codecs = + std::vector not_vp8_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) .codecs; @@ -1416,8 +1403,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, transceiver_or_error.MoveValue(); ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok()); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = vp8; RTCError error = video_transceiver->sender()->SetParameters(parameters); EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); @@ -1429,12 +1415,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional opus = + absl::optional opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - std::vector not_opus_codecs = + std::vector not_opus_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; @@ -1456,8 +1442,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = opus; RTCError error = audio_transceiver->sender()->SetParameters(parameters); EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); @@ -1469,12 +1454,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional opus = + absl::optional opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - std::vector not_opus_codecs = + std::vector not_opus_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; @@ -1519,8 +1504,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = opus; RTCError error = audio_transceiver->sender()->SetParameters(parameters); EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); @@ -1532,12 +1516,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional vp8 = + absl::optional vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - std::vector not_vp8_codecs = + std::vector not_vp8_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) .codecs; @@ -1559,8 +1543,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = vp8; RTCError error = video_transceiver->sender()->SetParameters(parameters); EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); @@ -1572,12 +1555,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional vp8 = + absl::optional vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - std::vector not_vp8_codecs = + std::vector not_vp8_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) .codecs; @@ -1622,8 +1605,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = vp8; RTCError error = video_transceiver->sender()->SetParameters(parameters); EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); @@ -1635,12 +1617,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional opus = + absl::optional opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - std::vector not_opus_codecs = + std::vector not_opus_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; @@ -1651,9 +1633,9 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, }), not_opus_codecs.end()); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = opus; init.send_encodings.push_back(encoding_parameters); @@ -1667,8 +1649,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); EXPECT_EQ(parameters.encodings[0].codec, opus); ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok()); @@ -1684,24 +1665,24 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - std::vector send_codecs = + std::vector send_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; - absl::optional opus = + absl::optional opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - absl::optional red = + absl::optional red = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "red"); ASSERT_TRUE(red); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = opus; init.send_encodings.push_back(encoding_parameters); @@ -1720,8 +1701,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); EXPECT_EQ(parameters.encodings[0].codec, opus); EXPECT_EQ(parameters.codecs[0].payload_type, red->preferred_payload_type); EXPECT_EQ(parameters.codecs[0].name, red->name); @@ -1743,14 +1723,14 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsScalabilityModeForSelectedCodec) { rtc::scoped_refptr local_pc_wrapper = CreatePc(); - absl::optional vp8 = + absl::optional vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = vp8; encoding_parameters.scalability_mode = "L1T3"; init.send_encodings.push_back(encoding_parameters); @@ -1761,8 +1741,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); parameters.encodings[0].scalability_mode = "L3T3"; RTCError error = video_transceiver->sender()->SetParameters(parameters); EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); @@ -1774,12 +1753,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional vp8 = + absl::optional vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - std::vector not_vp8_codecs = + std::vector not_vp8_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) .codecs; @@ -1790,9 +1769,9 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, }), not_vp8_codecs.end()); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.rid = "h"; encoding_parameters.codec = vp8; encoding_parameters.scale_resolution_down_by = 2; @@ -1811,8 +1790,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); ASSERT_EQ(parameters.encodings.size(), 2u); EXPECT_EQ(parameters.encodings[0].codec, vp8); EXPECT_EQ(parameters.encodings[1].codec, vp8); @@ -1833,17 +1811,17 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional vp8 = + absl::optional vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - absl::optional vp9 = + absl::optional vp9 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp9"); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.rid = "h"; encoding_parameters.codec = vp8; encoding_parameters.scale_resolution_down_by = 2; diff --git a/pc/peer_connection_factory_proxy.h b/pc/peer_connection_factory_proxy.h index 4781497642..b9bd1cbf0e 100644 --- a/pc/peer_connection_factory_proxy.h +++ b/pc/peer_connection_factory_proxy.h @@ -29,10 +29,10 @@ PROXY_METHOD2(RTCErrorOr>, CreatePeerConnectionOrError, const PeerConnectionInterface::RTCConfiguration&, PeerConnectionDependencies) -PROXY_CONSTMETHOD1(webrtc::RtpCapabilities, +PROXY_CONSTMETHOD1(RtpCapabilities, GetRtpSenderCapabilities, cricket::MediaType) -PROXY_CONSTMETHOD1(webrtc::RtpCapabilities, +PROXY_CONSTMETHOD1(RtpCapabilities, GetRtpReceiverCapabilities, cricket::MediaType) PROXY_METHOD1(rtc::scoped_refptr, diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc index 91772ec601..989b70f84e 100644 --- a/pc/peer_connection_factory_unittest.cc +++ b/pc/peer_connection_factory_unittest.cc @@ -106,8 +106,7 @@ class NullPeerConnectionObserver : public PeerConnectionObserver { PeerConnectionInterface::IceConnectionState new_state) override {} void OnIceGatheringChange( PeerConnectionInterface::IceGatheringState new_state) override {} - void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override { - } + void OnIceCandidate(const IceCandidateInterface* candidate) override {} }; class MockNetworkManager : public rtc::NetworkManager { @@ -133,17 +132,15 @@ class PeerConnectionFactoryTest : public ::testing::Test { private: void SetUp() { #ifdef WEBRTC_ANDROID - webrtc::InitializeAndroidObjects(); + InitializeAndroidObjects(); #endif // Use fake audio device module since we're only testing the interface // level, and using a real one could make tests flaky e.g. when run in // parallel. - factory_ = webrtc::CreatePeerConnectionFactory( + factory_ = CreatePeerConnectionFactory( rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(), - rtc::scoped_refptr( - FakeAudioCaptureModule::Create()), - webrtc::CreateBuiltinAudioEncoderFactory(), - webrtc::CreateBuiltinAudioDecoderFactory(), + rtc::scoped_refptr(FakeAudioCaptureModule::Create()), + CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(), std::make_unique>(), @@ -182,64 +179,64 @@ class PeerConnectionFactoryTest : public ::testing::Test { } } - void VerifyAudioCodecCapability(const webrtc::RtpCodecCapability& codec) { + void VerifyAudioCodecCapability(const RtpCodecCapability& codec) { EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_AUDIO); EXPECT_FALSE(codec.name.empty()); EXPECT_GT(codec.clock_rate, 0); EXPECT_GT(codec.num_channels, 0); } - void VerifyVideoCodecCapability(const webrtc::RtpCodecCapability& codec, + void VerifyVideoCodecCapability(const RtpCodecCapability& codec, bool sender) { EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_VIDEO); EXPECT_FALSE(codec.name.empty()); EXPECT_GT(codec.clock_rate, 0); if (sender) { if (codec.name == "VP8" || codec.name == "H264") { - EXPECT_THAT(codec.scalability_modes, - UnorderedElementsAre(webrtc::ScalabilityMode::kL1T1, - webrtc::ScalabilityMode::kL1T2, - webrtc::ScalabilityMode::kL1T3)) + EXPECT_THAT( + codec.scalability_modes, + UnorderedElementsAre(ScalabilityMode::kL1T1, ScalabilityMode::kL1T2, + ScalabilityMode::kL1T3)) << "Codec: " << codec.name; } else if (codec.name == "VP9" || codec.name == "AV1") { EXPECT_THAT( codec.scalability_modes, UnorderedElementsAre( // clang-format off - webrtc::ScalabilityMode::kL1T1, - webrtc::ScalabilityMode::kL1T2, - webrtc::ScalabilityMode::kL1T3, - webrtc::ScalabilityMode::kL2T1, - webrtc::ScalabilityMode::kL2T1h, - webrtc::ScalabilityMode::kL2T1_KEY, - webrtc::ScalabilityMode::kL2T2, - webrtc::ScalabilityMode::kL2T2h, - webrtc::ScalabilityMode::kL2T2_KEY, - webrtc::ScalabilityMode::kL2T2_KEY_SHIFT, - webrtc::ScalabilityMode::kL2T3, - webrtc::ScalabilityMode::kL2T3h, - webrtc::ScalabilityMode::kL2T3_KEY, - webrtc::ScalabilityMode::kL3T1, - webrtc::ScalabilityMode::kL3T1h, - webrtc::ScalabilityMode::kL3T1_KEY, - webrtc::ScalabilityMode::kL3T2, - webrtc::ScalabilityMode::kL3T2h, - webrtc::ScalabilityMode::kL3T2_KEY, - webrtc::ScalabilityMode::kL3T3, - webrtc::ScalabilityMode::kL3T3h, - webrtc::ScalabilityMode::kL3T3_KEY, - webrtc::ScalabilityMode::kS2T1, - webrtc::ScalabilityMode::kS2T1h, - webrtc::ScalabilityMode::kS2T2, - webrtc::ScalabilityMode::kS2T2h, - webrtc::ScalabilityMode::kS2T3, - webrtc::ScalabilityMode::kS2T3h, - webrtc::ScalabilityMode::kS3T1, - webrtc::ScalabilityMode::kS3T1h, - webrtc::ScalabilityMode::kS3T2, - webrtc::ScalabilityMode::kS3T2h, - webrtc::ScalabilityMode::kS3T3, - webrtc::ScalabilityMode::kS3T3h) + ScalabilityMode::kL1T1, + ScalabilityMode::kL1T2, + ScalabilityMode::kL1T3, + ScalabilityMode::kL2T1, + ScalabilityMode::kL2T1h, + ScalabilityMode::kL2T1_KEY, + ScalabilityMode::kL2T2, + ScalabilityMode::kL2T2h, + ScalabilityMode::kL2T2_KEY, + ScalabilityMode::kL2T2_KEY_SHIFT, + ScalabilityMode::kL2T3, + ScalabilityMode::kL2T3h, + ScalabilityMode::kL2T3_KEY, + ScalabilityMode::kL3T1, + ScalabilityMode::kL3T1h, + ScalabilityMode::kL3T1_KEY, + ScalabilityMode::kL3T2, + ScalabilityMode::kL3T2h, + ScalabilityMode::kL3T2_KEY, + ScalabilityMode::kL3T3, + ScalabilityMode::kL3T3h, + ScalabilityMode::kL3T3_KEY, + ScalabilityMode::kS2T1, + ScalabilityMode::kS2T1h, + ScalabilityMode::kS2T2, + ScalabilityMode::kS2T2h, + ScalabilityMode::kS2T3, + ScalabilityMode::kS2T3h, + ScalabilityMode::kS3T1, + ScalabilityMode::kS3T1h, + ScalabilityMode::kS3T2, + ScalabilityMode::kS3T2h, + ScalabilityMode::kS3T3, + ScalabilityMode::kS3T3h) // clang-format on ) << "Codec: " << codec.name; @@ -251,7 +248,7 @@ class PeerConnectionFactoryTest : public ::testing::Test { } } - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; std::unique_ptr socket_server_; rtc::AutoSocketServerThread main_thread_; rtc::scoped_refptr factory_; @@ -267,7 +264,7 @@ class PeerConnectionFactoryTest : public ::testing::Test { // to reconstruct factory with our own ConnectionContext. rtc::scoped_refptr CreatePeerConnectionFactoryWithRtxDisabled() { - webrtc::PeerConnectionFactoryDependencies pcf_dependencies; + PeerConnectionFactoryDependencies pcf_dependencies; pcf_dependencies.signaling_thread = rtc::Thread::Current(); pcf_dependencies.worker_thread = rtc::Thread::Current(); pcf_dependencies.network_thread = rtc::Thread::Current(); @@ -287,7 +284,7 @@ CreatePeerConnectionFactoryWithRtxDisabled() { OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(), EnableMedia(pcf_dependencies); - rtc::scoped_refptr context = + rtc::scoped_refptr context = ConnectionContext::Create(&pcf_dependencies); context->set_use_rtx(false); return rtc::make_ref_counted(context, @@ -302,26 +299,26 @@ CreatePeerConnectionFactoryWithRtxDisabled() { // See https://bugs.chromium.org/p/webrtc/issues/detail?id=7806 for details. TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) { #ifdef WEBRTC_ANDROID - webrtc::InitializeAndroidObjects(); + InitializeAndroidObjects(); #endif rtc::scoped_refptr factory( - webrtc::CreatePeerConnectionFactory( + CreatePeerConnectionFactory( nullptr /* network_thread */, nullptr /* worker_thread */, nullptr /* signaling_thread */, nullptr /* default_adm */, - webrtc::CreateBuiltinAudioEncoderFactory(), - webrtc::CreateBuiltinAudioDecoderFactory(), + CreateBuiltinAudioEncoderFactory(), + CreateBuiltinAudioDecoderFactory(), nullptr /* video_encoder_factory */, nullptr /* video_decoder_factory */, nullptr /* audio_mixer */, nullptr /* audio_processing */)); NullPeerConnectionObserver observer; - webrtc::PeerConnectionInterface::RTCConfiguration config; - config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; + PeerConnectionInterface::RTCConfiguration config; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; std::unique_ptr cert_generator( new FakeRTCCertificateGenerator()); - webrtc::PeerConnectionDependencies pc_dependencies(&observer); + PeerConnectionDependencies pc_dependencies(&observer); pc_dependencies.cert_generator = std::move(cert_generator); auto result = factory->CreatePeerConnectionOrError(config, std::move(pc_dependencies)); @@ -330,7 +327,7 @@ TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) { } TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) { - webrtc::RtpCapabilities audio_capabilities = + RtpCapabilities audio_capabilities = factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO); EXPECT_FALSE(audio_capabilities.codecs.empty()); for (const auto& codec : audio_capabilities.codecs) { @@ -343,7 +340,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) { } TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) { - webrtc::RtpCapabilities video_capabilities = + RtpCapabilities video_capabilities = factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO); EXPECT_FALSE(video_capabilities.codecs.empty()); for (const auto& codec : video_capabilities.codecs) { @@ -356,7 +353,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) { } TEST_F(PeerConnectionFactoryTest, CheckRtpSenderRtxEnabledCapabilities) { - webrtc::RtpCapabilities video_capabilities = + RtpCapabilities video_capabilities = factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO); const auto it = std::find_if( video_capabilities.codecs.begin(), video_capabilities.codecs.end(), @@ -366,7 +363,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpSenderRtxEnabledCapabilities) { TEST(PeerConnectionFactoryTestInternal, CheckRtpSenderRtxDisabledCapabilities) { auto factory = CreatePeerConnectionFactoryWithRtxDisabled(); - webrtc::RtpCapabilities video_capabilities = + RtpCapabilities video_capabilities = factory->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO); const auto it = std::find_if( video_capabilities.codecs.begin(), video_capabilities.codecs.end(), @@ -375,14 +372,14 @@ TEST(PeerConnectionFactoryTestInternal, CheckRtpSenderRtxDisabledCapabilities) { } TEST_F(PeerConnectionFactoryTest, CheckRtpSenderDataCapabilities) { - webrtc::RtpCapabilities data_capabilities = + RtpCapabilities data_capabilities = factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_DATA); EXPECT_TRUE(data_capabilities.codecs.empty()); EXPECT_TRUE(data_capabilities.header_extensions.empty()); } TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) { - webrtc::RtpCapabilities audio_capabilities = + RtpCapabilities audio_capabilities = factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO); EXPECT_FALSE(audio_capabilities.codecs.empty()); for (const auto& codec : audio_capabilities.codecs) { @@ -395,7 +392,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) { } TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) { - webrtc::RtpCapabilities video_capabilities = + RtpCapabilities video_capabilities = factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO); EXPECT_FALSE(video_capabilities.codecs.empty()); for (const auto& codec : video_capabilities.codecs) { @@ -408,7 +405,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) { } TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverRtxEnabledCapabilities) { - webrtc::RtpCapabilities video_capabilities = + RtpCapabilities video_capabilities = factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO); const auto it = std::find_if( video_capabilities.codecs.begin(), video_capabilities.codecs.end(), @@ -419,7 +416,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverRtxEnabledCapabilities) { TEST(PeerConnectionFactoryTestInternal, CheckRtpReceiverRtxDisabledCapabilities) { auto factory = CreatePeerConnectionFactoryWithRtxDisabled(); - webrtc::RtpCapabilities video_capabilities = + RtpCapabilities video_capabilities = factory->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO); const auto it = std::find_if( video_capabilities.codecs.begin(), video_capabilities.codecs.end(), @@ -428,7 +425,7 @@ TEST(PeerConnectionFactoryTestInternal, } TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) { - webrtc::RtpCapabilities data_capabilities = + RtpCapabilities data_capabilities = factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_DATA); EXPECT_TRUE(data_capabilities.codecs.empty()); EXPECT_TRUE(data_capabilities.header_extensions.empty()); @@ -438,8 +435,8 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) { // configuration. Also verifies the URL's parsed correctly as expected. TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) { PeerConnectionInterface::RTCConfiguration config; - config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; - webrtc::PeerConnectionInterface::IceServer ice_server; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; + PeerConnectionInterface::IceServer ice_server; ice_server.uri = kStunIceServer; config.servers.push_back(ice_server); ice_server.uri = kTurnIceServer; @@ -450,7 +447,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) { ice_server.username = kTurnUsername; ice_server.password = kTurnPassword; config.servers.push_back(ice_server); - webrtc::PeerConnectionDependencies pc_dependencies(&observer_); + PeerConnectionDependencies pc_dependencies(&observer_); pc_dependencies.cert_generator = std::make_unique(); pc_dependencies.allocator = std::move(port_allocator_); @@ -475,15 +472,15 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) { // configuration. Also verifies the list of URL's parsed correctly as expected. TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) { PeerConnectionInterface::RTCConfiguration config; - config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; - webrtc::PeerConnectionInterface::IceServer ice_server; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; + PeerConnectionInterface::IceServer ice_server; ice_server.urls.push_back(kStunIceServer); ice_server.urls.push_back(kTurnIceServer); ice_server.urls.push_back(kTurnIceServerWithTransport); ice_server.username = kTurnUsername; ice_server.password = kTurnPassword; config.servers.push_back(ice_server); - webrtc::PeerConnectionDependencies pc_dependencies(&observer_); + PeerConnectionDependencies pc_dependencies(&observer_); pc_dependencies.cert_generator = std::make_unique(); pc_dependencies.allocator = std::move(port_allocator_); @@ -506,15 +503,15 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) { TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) { PeerConnectionInterface::RTCConfiguration config; - config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; - webrtc::PeerConnectionInterface::IceServer ice_server; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; + PeerConnectionInterface::IceServer ice_server; ice_server.uri = kStunIceServer; config.servers.push_back(ice_server); ice_server.uri = kTurnIceServerWithNoUsernameInUri; ice_server.username = kTurnUsername; ice_server.password = kTurnPassword; config.servers.push_back(ice_server); - webrtc::PeerConnectionDependencies pc_dependencies(&observer_); + PeerConnectionDependencies pc_dependencies(&observer_); pc_dependencies.cert_generator = std::make_unique(); pc_dependencies.allocator = std::move(port_allocator_); @@ -532,13 +529,13 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) { // has transport parameter in it. TEST_F(PeerConnectionFactoryTest, CreatePCUsingTurnUrlWithTransportParam) { PeerConnectionInterface::RTCConfiguration config; - config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; - webrtc::PeerConnectionInterface::IceServer ice_server; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; + PeerConnectionInterface::IceServer ice_server; ice_server.uri = kTurnIceServerWithTransport; ice_server.username = kTurnUsername; ice_server.password = kTurnPassword; config.servers.push_back(ice_server); - webrtc::PeerConnectionDependencies pc_dependencies(&observer_); + PeerConnectionDependencies pc_dependencies(&observer_); pc_dependencies.cert_generator = std::make_unique(); pc_dependencies.allocator = std::move(port_allocator_); @@ -554,8 +551,8 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingTurnUrlWithTransportParam) { TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) { PeerConnectionInterface::RTCConfiguration config; - config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; - webrtc::PeerConnectionInterface::IceServer ice_server; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; + PeerConnectionInterface::IceServer ice_server; ice_server.uri = kSecureTurnIceServer; ice_server.username = kTurnUsername; ice_server.password = kTurnPassword; @@ -568,7 +565,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) { ice_server.username = kTurnUsername; ice_server.password = kTurnPassword; config.servers.push_back(ice_server); - webrtc::PeerConnectionDependencies pc_dependencies(&observer_); + PeerConnectionDependencies pc_dependencies(&observer_); pc_dependencies.cert_generator = std::make_unique(); pc_dependencies.allocator = std::move(port_allocator_); @@ -593,8 +590,8 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) { TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) { PeerConnectionInterface::RTCConfiguration config; - config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; - webrtc::PeerConnectionInterface::IceServer ice_server; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; + PeerConnectionInterface::IceServer ice_server; ice_server.uri = kStunIceServerWithIPv4Address; config.servers.push_back(ice_server); ice_server.uri = kStunIceServerWithIPv4AddressWithoutPort; @@ -607,7 +604,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) { ice_server.username = kTurnUsername; ice_server.password = kTurnPassword; config.servers.push_back(ice_server); - webrtc::PeerConnectionDependencies pc_dependencies(&observer_); + PeerConnectionDependencies pc_dependencies(&observer_); pc_dependencies.cert_generator = std::make_unique(); pc_dependencies.allocator = std::move(port_allocator_); @@ -635,8 +632,8 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) { // This test verifies the captured stream is rendered locally using a // local video track. TEST_F(PeerConnectionFactoryTest, LocalRendering) { - rtc::scoped_refptr source = - webrtc::FakeVideoTrackSource::Create(/*is_screencast=*/false); + rtc::scoped_refptr source = + FakeVideoTrackSource::Create(/*is_screencast=*/false); cricket::FakeFrameSource frame_source(1280, 720, rtc::kNumMicrosecsPerSec / 30); @@ -664,7 +661,7 @@ TEST_F(PeerConnectionFactoryTest, LocalRendering) { } TEST(PeerConnectionFactoryDependenciesTest, UsesNetworkManager) { - constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10); + constexpr TimeDelta kWaitTimeout = TimeDelta::Seconds(10); auto mock_network_manager = std::make_unique>(); rtc::Event called; @@ -672,24 +669,24 @@ TEST(PeerConnectionFactoryDependenciesTest, UsesNetworkManager) { .Times(AtLeast(1)) .WillRepeatedly(InvokeWithoutArgs([&] { called.Set(); })); - webrtc::PeerConnectionFactoryDependencies pcf_dependencies; + PeerConnectionFactoryDependencies pcf_dependencies; pcf_dependencies.network_manager = std::move(mock_network_manager); - rtc::scoped_refptr pcf = + rtc::scoped_refptr pcf = CreateModularPeerConnectionFactory(std::move(pcf_dependencies)); PeerConnectionInterface::RTCConfiguration config; config.ice_candidate_pool_size = 2; NullPeerConnectionObserver observer; auto pc = pcf->CreatePeerConnectionOrError( - config, webrtc::PeerConnectionDependencies(&observer)); + config, PeerConnectionDependencies(&observer)); ASSERT_TRUE(pc.ok()); called.Wait(kWaitTimeout); } TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) { - constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10); + constexpr TimeDelta kWaitTimeout = TimeDelta::Seconds(10); auto mock_socket_factory = std::make_unique>(); @@ -701,10 +698,10 @@ TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) { })) .WillRepeatedly(Return(nullptr)); - webrtc::PeerConnectionFactoryDependencies pcf_dependencies; + PeerConnectionFactoryDependencies pcf_dependencies; pcf_dependencies.packet_socket_factory = std::move(mock_socket_factory); - rtc::scoped_refptr pcf = + rtc::scoped_refptr pcf = CreateModularPeerConnectionFactory(std::move(pcf_dependencies)); // By default, localhost addresses are ignored, which makes tests fail if test @@ -717,7 +714,7 @@ TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) { config.ice_candidate_pool_size = 2; NullPeerConnectionObserver observer; auto pc = pcf->CreatePeerConnectionOrError( - config, webrtc::PeerConnectionDependencies(&observer)); + config, PeerConnectionDependencies(&observer)); ASSERT_TRUE(pc.ok()); called.Wait(kWaitTimeout); diff --git a/pc/peer_connection_field_trial_tests.cc b/pc/peer_connection_field_trial_tests.cc index c009475c7e..4cbe24986c 100644 --- a/pc/peer_connection_field_trial_tests.cc +++ b/pc/peer_connection_field_trial_tests.cc @@ -68,7 +68,7 @@ class PeerConnectionFieldTrialTest : public ::testing::Test { #ifdef WEBRTC_ANDROID InitializeAndroidObjects(); #endif - webrtc::PeerConnectionInterface::IceServer ice_server; + PeerConnectionInterface::IceServer ice_server; ice_server.uri = "stun:stun.l.google.com:19302"; config_.servers.push_back(ice_server); config_.sdp_semantics = SdpSemantics::kUnifiedPlan; @@ -108,7 +108,7 @@ class PeerConnectionFieldTrialTest : public ::testing::Test { std::unique_ptr socket_server_; rtc::AutoSocketServerThread main_thread_; rtc::scoped_refptr pc_factory_ = nullptr; - webrtc::PeerConnectionInterface::RTCConfiguration config_; + PeerConnectionInterface::RTCConfiguration config_; }; // Tests for the dependency descriptor field trial. The dependency descriptor @@ -133,7 +133,7 @@ TEST_F(PeerConnectionFieldTrialTest, EnableDependencyDescriptorAdvertised) { media_description1->rtp_header_extensions(); bool found = absl::c_find_if(rtp_header_extensions1, - [](const webrtc::RtpExtension& rtp_extension) { + [](const RtpExtension& rtp_extension) { return rtp_extension.uri == RtpExtension::kDependencyDescriptorUri; }) != rtp_header_extensions1.end(); @@ -163,14 +163,14 @@ TEST_F(PeerConnectionFieldTrialTest, InjectDependencyDescriptor) { media_description1->rtp_header_extensions(); bool found1 = absl::c_find_if(rtp_header_extensions1, - [](const webrtc::RtpExtension& rtp_extension) { + [](const RtpExtension& rtp_extension) { return rtp_extension.uri == RtpExtension::kDependencyDescriptorUri; }) != rtp_header_extensions1.end(); EXPECT_FALSE(found1); std::set existing_ids; - for (const webrtc::RtpExtension& rtp_extension : rtp_header_extensions1) { + for (const RtpExtension& rtp_extension : rtp_header_extensions1) { existing_ids.insert(rtp_extension.id); } @@ -207,7 +207,7 @@ TEST_F(PeerConnectionFieldTrialTest, InjectDependencyDescriptor) { media_description2->rtp_header_extensions(); bool found2 = absl::c_find_if(rtp_header_extensions2, - [](const webrtc::RtpExtension& rtp_extension) { + [](const RtpExtension& rtp_extension) { return rtp_extension.uri == RtpExtension::kDependencyDescriptorUri; }) != rtp_header_extensions2.end(); diff --git a/pc/peer_connection_header_extension_unittest.cc b/pc/peer_connection_header_extension_unittest.cc index dd5d4b097a..277979b330 100644 --- a/pc/peer_connection_header_extension_unittest.cc +++ b/pc/peer_connection_header_extension_unittest.cc @@ -114,7 +114,7 @@ class PeerConnectionHeaderExtensionTest pc_factory, result.MoveValue(), std::move(observer)); } - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; std::unique_ptr socket_server_; rtc::AutoSocketServerThread main_thread_; std::vector extensions_; diff --git a/pc/peer_connection_histogram_unittest.cc b/pc/peer_connection_histogram_unittest.cc index cadb0839ba..973744c0e8 100644 --- a/pc/peer_connection_histogram_unittest.cc +++ b/pc/peer_connection_histogram_unittest.cc @@ -94,7 +94,7 @@ typedef PeerConnectionWrapperForUsageHistogramTest* RawWrapperPtr; class ObserverForUsageHistogramTest : public MockPeerConnectionObserver { public: - void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; + void OnIceCandidate(const IceCandidateInterface* candidate) override; void OnInterestingUsage(int usage_pattern) override { interesting_usage_detected_ = usage_pattern; @@ -157,12 +157,11 @@ class PeerConnectionWrapperForUsageHistogramTest return static_cast(observer()) ->HaveDataChannel(); } - void BufferIceCandidate(const webrtc::IceCandidateInterface* candidate) { + void BufferIceCandidate(const IceCandidateInterface* candidate) { std::string sdp; EXPECT_TRUE(candidate->ToString(&sdp)); - std::unique_ptr candidate_copy( - CreateIceCandidate(candidate->sdp_mid(), candidate->sdp_mline_index(), - sdp, nullptr)); + std::unique_ptr candidate_copy(CreateIceCandidate( + candidate->sdp_mid(), candidate->sdp_mline_index(), sdp, nullptr)); buffered_candidates_.push_back(std::move(candidate_copy)); } @@ -213,19 +212,18 @@ class PeerConnectionWrapperForUsageHistogramTest return true; } - webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() { + PeerConnectionInterface::IceGatheringState ice_gathering_state() { return pc()->ice_gathering_state(); } private: // Candidates that have been sent but not yet configured - std::vector> - buffered_candidates_; + std::vector> buffered_candidates_; }; // Buffers candidates until we add them via AddBufferedIceCandidates. void ObserverForUsageHistogramTest::OnIceCandidate( - const webrtc::IceCandidateInterface* candidate) { + const IceCandidateInterface* candidate) { // If target is not set, ignore. This happens in one-ended unit tests. if (candidate_target_) { this->candidate_target_->BufferIceCandidate(candidate); @@ -242,12 +240,12 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test { : vss_(new rtc::VirtualSocketServer()), socket_factory_(new rtc::BasicPacketSocketFactory(vss_.get())), main_(vss_.get()) { - webrtc::metrics::Reset(); + metrics::Reset(); } WrapperPtr CreatePeerConnection() { RTCConfiguration config; - config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; return CreatePeerConnection( config, PeerConnectionFactoryInterface::Options(), nullptr); } @@ -259,13 +257,13 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test { WrapperPtr CreatePeerConnectionWithMdns(const RTCConfiguration& config) { auto resolver_factory = - std::make_unique>(); + std::make_unique>(); - webrtc::PeerConnectionDependencies deps(nullptr /* observer_in */); + PeerConnectionDependencies deps(nullptr /* observer_in */); auto fake_network = NewFakeNetwork(); fake_network->set_mdns_responder( - std::make_unique(rtc::Thread::Current())); + std::make_unique(rtc::Thread::Current())); fake_network->AddInterface(NextLocalAddress()); std::unique_ptr port_allocator( @@ -280,7 +278,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test { WrapperPtr CreatePeerConnectionWithImmediateReport() { RTCConfiguration configuration; - configuration.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; + configuration.sdp_semantics = SdpSemantics::kUnifiedPlan; configuration.report_usage_pattern_delay_ms = 0; return CreatePeerConnection( configuration, PeerConnectionFactoryInterface::Options(), nullptr); @@ -361,7 +359,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test { // This works correctly only if there is only one sample value // that has been counted. // Returns -1 for "not found". - return webrtc::metrics::MinSample(kUsagePatternMetric); + return metrics::MinSample(kUsagePatternMetric); } // The PeerConnection's port allocator is tied to the PeerConnection's @@ -390,10 +388,10 @@ TEST_F(PeerConnectionUsageHistogramTest, UsageFingerprintHistogramFromTimeout) { auto pc = CreatePeerConnectionWithImmediateReport(); int expected_fingerprint = MakeUsageFingerprint({}); - EXPECT_METRIC_EQ_WAIT(1, webrtc::metrics::NumSamples(kUsagePatternMetric), + EXPECT_METRIC_EQ_WAIT(1, metrics::NumSamples(kUsagePatternMetric), kDefaultTimeout); EXPECT_METRIC_EQ( - 1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint)); + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint)); } #ifndef WEBRTC_ANDROID @@ -418,11 +416,10 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintAudioVideo) { UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED}); // In this case, we may or may not have PRIVATE_CANDIDATE_COLLECTED, // depending on the machine configuration. - EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric)); EXPECT_METRIC_TRUE( - webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) == - 2 || - webrtc::metrics::NumEvents( + metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) == 2 || + metrics::NumEvents( kUsagePatternMetric, expected_fingerprint | static_cast(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) == 2); @@ -463,11 +460,11 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithMdnsCaller) { UsageEvent::CANDIDATE_COLLECTED, UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED, UsageEvent::REMOTE_MDNS_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED, UsageEvent::REMOTE_CANDIDATE_ADDED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_caller)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_callee)); + EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee)); } // Test getting the usage fingerprint when the callee collects an mDNS @@ -504,11 +501,11 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithMdnsCallee) { UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED, UsageEvent::ICE_STATE_CONNECTED, UsageEvent::REMOTE_CANDIDATE_ADDED, UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_caller)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_callee)); + EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee)); } #ifdef WEBRTC_HAVE_SCTP @@ -526,11 +523,10 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) { UsageEvent::CANDIDATE_COLLECTED, UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED, UsageEvent::ICE_STATE_CONNECTED, UsageEvent::REMOTE_CANDIDATE_ADDED, UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric)); EXPECT_METRIC_TRUE( - webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) == - 2 || - webrtc::metrics::NumEvents( + metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) == 2 || + metrics::NumEvents( kUsagePatternMetric, expected_fingerprint | static_cast(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) == 2); @@ -554,9 +550,9 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurn) { int expected_fingerprint = MakeUsageFingerprint( {UsageEvent::STUN_SERVER_ADDED, UsageEvent::TURN_SERVER_ADDED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric)); EXPECT_METRIC_EQ( - 1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint)); + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint)); } TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) { @@ -576,9 +572,9 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) { int expected_fingerprint = MakeUsageFingerprint( {UsageEvent::STUN_SERVER_ADDED, UsageEvent::TURN_SERVER_ADDED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric)); EXPECT_METRIC_EQ( - 1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint)); + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint)); } TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIPCaller) { @@ -604,11 +600,11 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIPCaller) { UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED, UsageEvent::REMOTE_CANDIDATE_ADDED, UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_caller)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_callee)); + EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee)); } TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIpv6Callee) { @@ -636,11 +632,11 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIpv6Callee) { UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED, UsageEvent::REMOTE_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED, UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_caller)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_callee)); + EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee)); } #ifndef WEBRTC_ANDROID @@ -664,7 +660,7 @@ TEST_F(PeerConnectionUsageHistogramTest, ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer())); // Wait until the gathering completes so that the session description would // have contained ICE candidates. - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete, caller->ice_gathering_state(), kDefaultTimeout); EXPECT_TRUE(caller->observer()->candidate_gathered()); // Get the current offer that contains candidates and pass it to the callee. @@ -713,11 +709,11 @@ TEST_F(PeerConnectionUsageHistogramTest, UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED, UsageEvent::REMOTE_IPV6_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED, UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_caller)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric, - expected_fingerprint_callee)); + EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee)); } TEST_F(PeerConnectionUsageHistogramTest, NotableUsageNoted) { @@ -728,7 +724,7 @@ TEST_F(PeerConnectionUsageHistogramTest, NotableUsageNoted) { int expected_fingerprint = MakeUsageFingerprint( {UsageEvent::DATA_ADDED, UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED, UsageEvent::CANDIDATE_COLLECTED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric)); EXPECT_METRIC_TRUE( expected_fingerprint == ObservedFingerprint() || (expected_fingerprint | @@ -745,9 +741,9 @@ TEST_F(PeerConnectionUsageHistogramTest, NotableUsageOnEventFiring) { int expected_fingerprint = MakeUsageFingerprint( {UsageEvent::DATA_ADDED, UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED, UsageEvent::CANDIDATE_COLLECTED}); - EXPECT_METRIC_EQ(0, webrtc::metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ(0, metrics::NumSamples(kUsagePatternMetric)); caller->GetInternalPeerConnection()->RequestUsagePatternReportForTesting(); - EXPECT_METRIC_EQ_WAIT(1, webrtc::metrics::NumSamples(kUsagePatternMetric), + EXPECT_METRIC_EQ_WAIT(1, metrics::NumSamples(kUsagePatternMetric), kDefaultTimeout); EXPECT_METRIC_TRUE( expected_fingerprint == ObservedFingerprint() || @@ -766,12 +762,12 @@ TEST_F(PeerConnectionUsageHistogramTest, int expected_fingerprint = MakeUsageFingerprint( {UsageEvent::DATA_ADDED, UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED, UsageEvent::CANDIDATE_COLLECTED, UsageEvent::CLOSE_CALLED}); - EXPECT_METRIC_EQ(0, webrtc::metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ(0, metrics::NumSamples(kUsagePatternMetric)); caller->pc()->Close(); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric)); + EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric)); caller->GetInternalPeerConnection()->RequestUsagePatternReportForTesting(); caller->observer()->ClearInterestingUsageDetector(); - EXPECT_METRIC_EQ_WAIT(2, webrtc::metrics::NumSamples(kUsagePatternMetric), + EXPECT_METRIC_EQ_WAIT(2, metrics::NumSamples(kUsagePatternMetric), kDefaultTimeout); EXPECT_METRIC_TRUE( expected_fingerprint == ObservedFingerprint() || diff --git a/pc/peer_connection_ice_unittest.cc b/pc/peer_connection_ice_unittest.cc index 532583f307..492e108cbc 100644 --- a/pc/peer_connection_ice_unittest.cc +++ b/pc/peer_connection_ice_unittest.cc @@ -342,7 +342,7 @@ class PeerConnectionIceTest public ::testing::WithParamInterface { protected: PeerConnectionIceTest() : PeerConnectionIceBaseTest(GetParam()) { - webrtc::metrics::Reset(); + metrics::Reset(); } }; @@ -514,7 +514,7 @@ TEST_P(PeerConnectionIceTest, CannotAddCandidateWhenRemoteDescriptionNotSet) { EXPECT_FALSE(caller->pc()->AddIceCandidate(jsep_candidate.get())); EXPECT_METRIC_THAT( - webrtc::metrics::Samples("WebRTC.PeerConnection.AddIceCandidate"), + metrics::Samples("WebRTC.PeerConnection.AddIceCandidate"), ElementsAre(Pair(kAddIceCandidateFailNoRemoteDescription, 2))); } @@ -1457,7 +1457,7 @@ class PeerConnectionIceConfigTest : public ::testing::Test { pc_ = result.MoveValue(); } - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; std::unique_ptr socket_server_; rtc::AutoSocketServerThread main_thread_; rtc::scoped_refptr pc_factory_ = nullptr; diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc index d76e5e27d5..1ea16a6cf9 100644 --- a/pc/peer_connection_integrationtest.cc +++ b/pc/peer_connection_integrationtest.cc @@ -124,7 +124,7 @@ class FakeClockForTest : public rtc::ScopedFakeClock { // Some things use a time of "0" as a special value, so we need to start out // the fake clock at a nonzero time. // TODO(deadbeef): Fix this. - AdvanceTime(webrtc::TimeDelta::Seconds(1)); + AdvanceTime(TimeDelta::Seconds(1)); } // Explicit handle. @@ -324,7 +324,7 @@ TEST_P(PeerConnectionIntegrationTest, ConnectFakeSignaling(); // Add video tracks with 16:9 aspect ratio, size 1280 x 720. - webrtc::FakePeriodicVideoSource::Config config; + FakePeriodicVideoSource::Config config; config.width = 1280; config.height = 720; config.timestamp_offset_ms = rtc::TimeMillis(); @@ -366,7 +366,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSendOnlyVideo) { CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/true)); ConnectFakeSignaling(); // Add one-directional video, from caller to callee. - rtc::scoped_refptr caller_track = + rtc::scoped_refptr caller_track = caller()->CreateLocalVideoTrack(); caller()->AddTrack(caller_track); PeerConnectionInterface::RTCOfferAnswerOptions options; @@ -391,7 +391,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithReceiveOnlyVideo) { CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/false)); ConnectFakeSignaling(); // Add one-directional video, from callee to caller. - rtc::scoped_refptr callee_track = + rtc::scoped_refptr callee_track = callee()->CreateLocalVideoTrack(); callee()->AddTrack(callee_track); PeerConnectionInterface::RTCOfferAnswerOptions options; @@ -414,14 +414,14 @@ TEST_P(PeerConnectionIntegrationTest, ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); // Add one-directional video, from caller to callee. - rtc::scoped_refptr caller_track = + rtc::scoped_refptr caller_track = caller()->CreateLocalVideoTrack(); caller()->AddTrack(caller_track); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); // Add receive video. - rtc::scoped_refptr callee_track = + rtc::scoped_refptr callee_track = callee()->CreateLocalVideoTrack(); callee()->AddTrack(callee_track); caller()->CreateAndSetAndSignalOffer(); @@ -438,14 +438,14 @@ TEST_P(PeerConnectionIntegrationTest, ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); // Add one-directional video, from callee to caller. - rtc::scoped_refptr callee_track = + rtc::scoped_refptr callee_track = callee()->CreateLocalVideoTrack(); callee()->AddTrack(callee_track); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); // Add send video. - rtc::scoped_refptr caller_track = + rtc::scoped_refptr caller_track = caller()->CreateLocalVideoTrack(); caller()->AddTrack(caller_track); caller()->CreateAndSetAndSignalOffer(); @@ -462,15 +462,15 @@ TEST_P(PeerConnectionIntegrationTest, ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); // Add send video, from caller to callee. - rtc::scoped_refptr caller_track = + rtc::scoped_refptr caller_track = caller()->CreateLocalVideoTrack(); - rtc::scoped_refptr caller_sender = + rtc::scoped_refptr caller_sender = caller()->AddTrack(caller_track); // Add receive video, from callee to caller. - rtc::scoped_refptr callee_track = + rtc::scoped_refptr callee_track = callee()->CreateLocalVideoTrack(); - rtc::scoped_refptr callee_sender = + rtc::scoped_refptr callee_sender = callee()->AddTrack(callee_track); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); @@ -494,15 +494,15 @@ TEST_P(PeerConnectionIntegrationTest, ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); // Add send video, from caller to callee. - rtc::scoped_refptr caller_track = + rtc::scoped_refptr caller_track = caller()->CreateLocalVideoTrack(); - rtc::scoped_refptr caller_sender = + rtc::scoped_refptr caller_sender = caller()->AddTrack(caller_track); // Add receive video, from callee to caller. - rtc::scoped_refptr callee_track = + rtc::scoped_refptr callee_track = callee()->CreateLocalVideoTrack(); - rtc::scoped_refptr callee_sender = + rtc::scoped_refptr callee_sender = callee()->AddTrack(callee_track); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); @@ -654,9 +654,9 @@ TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) { ConnectFakeSignaling(); // Add rotated video tracks. caller()->AddTrack( - caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90)); + caller()->CreateLocalVideoTrackWithRotation(kVideoRotation_90)); callee()->AddTrack( - callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270)); + callee()->CreateLocalVideoTrackWithRotation(kVideoRotation_270)); // Wait for video frames to be received by both sides. caller()->CreateAndSetAndSignalOffer(); @@ -673,8 +673,8 @@ TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) { EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio()); EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio()); // Ensure that the CVO bits were surfaced to the renderer. - EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation()); - EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation()); + EXPECT_EQ(kVideoRotation_270, caller()->rendered_rotation()); + EXPECT_EQ(kVideoRotation_90, callee()->rendered_rotation()); } // Test that when the CVO extension isn't supported, video is rotated the @@ -684,9 +684,9 @@ TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) { ConnectFakeSignaling(); // Add rotated video tracks. caller()->AddTrack( - caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90)); + caller()->CreateLocalVideoTrackWithRotation(kVideoRotation_90)); callee()->AddTrack( - callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270)); + callee()->CreateLocalVideoTrackWithRotation(kVideoRotation_270)); // Remove the CVO extension from the offered SDP. callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) { @@ -710,8 +710,8 @@ TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) { EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio()); EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio()); // Expect that each endpoint is unaware of the rotation of the other endpoint. - EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation()); - EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation()); + EXPECT_EQ(kVideoRotation_0, caller()->rendered_rotation()); + EXPECT_EQ(kVideoRotation_0, callee()->rendered_rotation()); } // Test that if the answerer rejects the audio m= section, no audio is sent or @@ -899,9 +899,9 @@ TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) { ConnectFakeSignaling(); // Add audio track, do normal offer/answer. - rtc::scoped_refptr track = + rtc::scoped_refptr track = caller()->CreateLocalAudioTrack(); - rtc::scoped_refptr sender = + rtc::scoped_refptr sender = caller()->pc()->AddTrack(track, {"stream"}).MoveValue(); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); @@ -974,7 +974,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); // Add one-directional video, from caller to callee. - rtc::scoped_refptr track = + rtc::scoped_refptr track = caller()->CreateLocalVideoTrack(); RtpTransceiverInit video_transceiver_init; @@ -988,7 +988,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, // Add receive direction. video_sender->SetDirectionWithError(RtpTransceiverDirection::kSendRecv); - rtc::scoped_refptr callee_track = + rtc::scoped_refptr callee_track = callee()->CreateLocalVideoTrack(); callee()->AddTrack(callee_track); @@ -1348,11 +1348,11 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) { audio_sender_1->track()->id(), video_sender_1->track()->id(), audio_sender_2->track()->id(), video_sender_2->track()->id()}; - rtc::scoped_refptr caller_report = + rtc::scoped_refptr caller_report = caller()->NewGetStats(); ASSERT_TRUE(caller_report); auto outbound_stream_stats = - caller_report->GetStatsOfType(); + caller_report->GetStatsOfType(); ASSERT_EQ(outbound_stream_stats.size(), 4u); std::vector outbound_track_ids; for (const auto& stat : outbound_stream_stats) { @@ -1373,11 +1373,11 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) { } EXPECT_THAT(outbound_track_ids, UnorderedElementsAreArray(track_ids)); - rtc::scoped_refptr callee_report = + rtc::scoped_refptr callee_report = callee()->NewGetStats(); ASSERT_TRUE(callee_report); auto inbound_stream_stats = - callee_report->GetStatsOfType(); + callee_report->GetStatsOfType(); ASSERT_EQ(4u, inbound_stream_stats.size()); std::vector inbound_track_ids; for (const auto& stat : inbound_stream_stats) { @@ -1412,11 +1412,10 @@ TEST_P(PeerConnectionIntegrationTest, // We received a frame, so we should have nonzero "bytes received" stats for // the unsignaled stream, if stats are working for it. - rtc::scoped_refptr report = - callee()->NewGetStats(); + rtc::scoped_refptr report = callee()->NewGetStats(); ASSERT_NE(nullptr, report); auto inbound_stream_stats = - report->GetStatsOfType(); + report->GetStatsOfType(); ASSERT_EQ(1U, inbound_stream_stats.size()); ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined()); ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U); @@ -1459,12 +1458,10 @@ TEST_P(PeerConnectionIntegrationTest, media_expectations.CalleeExpectsSomeVideo(1); ASSERT_TRUE(ExpectNewFrames(media_expectations)); - rtc::scoped_refptr report = - callee()->NewGetStats(); + rtc::scoped_refptr report = callee()->NewGetStats(); ASSERT_NE(nullptr, report); - auto inbound_rtps = - report->GetStatsOfType(); + auto inbound_rtps = report->GetStatsOfType(); auto index = FindFirstMediaStatsIndexByKind("audio", inbound_rtps); ASSERT_GE(index, 0); EXPECT_TRUE(inbound_rtps[index]->audio_level.is_defined()); @@ -1655,18 +1652,18 @@ TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) { callee()->AddAudioVideoTracks(); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete, caller()->ice_gathering_state(), kMaxWaitForFramesMs); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete, callee()->ice_gathering_state(), kMaxWaitForFramesMs); // After the best candidate pair is selected and all candidates are signaled, // the ICE connection state should reach "complete". // TODO(deadbeef): Currently, the ICE "controlled" agent (the // answerer/"callee" by default) only reaches "connected". When this is // fixed, this test should be updated. - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, caller()->ice_connection_state(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, callee()->ice_connection_state(), kDefaultTimeout); } @@ -1679,9 +1676,9 @@ constexpr int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN | TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletionWithRemoteHostname) { auto caller_resolver_factory = - std::make_unique>(); + std::make_unique>(); auto callee_resolver_factory = - std::make_unique>(); + std::make_unique>(); auto callee_async_resolver = std::make_unique>(); auto caller_async_resolver = @@ -1695,12 +1692,12 @@ TEST_P(PeerConnectionIntegrationTest, // P2PTransportChannel. EXPECT_CALL(*caller_resolver_factory, Create()) .WillOnce(Return(ByMove(std::move(caller_async_resolver)))); - webrtc::PeerConnectionDependencies caller_deps(nullptr); + PeerConnectionDependencies caller_deps(nullptr); caller_deps.async_dns_resolver_factory = std::move(caller_resolver_factory); EXPECT_CALL(*callee_resolver_factory, Create()) .WillOnce(Return(ByMove(std::move(callee_async_resolver)))); - webrtc::PeerConnectionDependencies callee_deps(nullptr); + PeerConnectionDependencies callee_deps(nullptr); callee_deps.async_dns_resolver_factory = std::move(callee_resolver_factory); PeerConnectionInterface::RTCConfiguration config; @@ -1719,9 +1716,9 @@ TEST_P(PeerConnectionIntegrationTest, // Enable hostname candidates with mDNS names. caller()->SetMdnsResponder( - std::make_unique(network_thread())); + std::make_unique(network_thread())); callee()->SetMdnsResponder( - std::make_unique(network_thread())); + std::make_unique(network_thread())); SetPortAllocatorFlags(kOnlyLocalPorts, kOnlyLocalPorts); @@ -1730,18 +1727,18 @@ TEST_P(PeerConnectionIntegrationTest, callee()->AddAudioVideoTracks(); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, caller()->ice_connection_state(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, callee()->ice_connection_state(), kDefaultTimeout); // Part of reporting the stats will occur on the network thread, so flush it // before checking NumEvents. SendTask(network_thread(), [] {}); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents( - "WebRTC.PeerConnection.CandidatePairType_UDP", - webrtc::kIceCandidatePairHostNameHostName)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP", + kIceCandidatePairHostNameHostName)); DestroyPeerConnections(); } @@ -1862,9 +1859,9 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_VerifyBestConnection) { caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, caller()->ice_connection_state(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, callee()->ice_connection_state(), kDefaultTimeout); // Part of reporting the stats will occur on the network thread, so flush it @@ -1872,10 +1869,10 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_VerifyBestConnection) { SendTask(network_thread(), [] {}); // TODO(bugs.webrtc.org/9456): Fix it. - const int num_best_ipv4 = webrtc::metrics::NumEvents( - "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4); - const int num_best_ipv6 = webrtc::metrics::NumEvents( - "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6); + const int num_best_ipv4 = metrics::NumEvents( + "WebRTC.PeerConnection.IPMetrics", kBestConnections_IPv4); + const int num_best_ipv6 = metrics::NumEvents( + "WebRTC.PeerConnection.IPMetrics", kBestConnections_IPv6); if (TestIPv6()) { // When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4 // connection. @@ -1886,12 +1883,12 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_VerifyBestConnection) { EXPECT_METRIC_EQ(0, num_best_ipv6); } - EXPECT_METRIC_EQ(0, webrtc::metrics::NumEvents( - "WebRTC.PeerConnection.CandidatePairType_UDP", - webrtc::kIceCandidatePairHostHost)); - EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents( - "WebRTC.PeerConnection.CandidatePairType_UDP", - webrtc::kIceCandidatePairHostPublicHostPublic)); + EXPECT_METRIC_EQ( + 0, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP", + kIceCandidatePairHostHost)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP", + kIceCandidatePairHostPublicHostPublic)); } constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP | @@ -1931,17 +1928,17 @@ TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) { callee()->AddAudioVideoTracks(); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, caller()->ice_connection_state(), kMaxWaitForFramesMs); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, callee()->ice_connection_state(), kMaxWaitForFramesMs); // To verify that the ICE restart actually occurs, get // ufrag/password/candidates before and after restart. // Create an SDP string of the first audio candidate for both clients. - const webrtc::IceCandidateCollection* audio_candidates_caller = + const IceCandidateCollection* audio_candidates_caller = caller()->pc()->local_description()->candidates(0); - const webrtc::IceCandidateCollection* audio_candidates_callee = + const IceCandidateCollection* audio_candidates_callee = callee()->pc()->local_description()->candidates(0); ASSERT_GT(audio_candidates_caller->count(), 0u); ASSERT_GT(audio_candidates_callee->count(), 0u); @@ -1964,9 +1961,9 @@ TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) { caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, caller()->ice_connection_state(), kMaxWaitForFramesMs); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, callee()->ice_connection_state(), kMaxWaitForFramesMs); // Grab the ufrags/candidates again. @@ -2141,9 +2138,9 @@ TEST_F(PeerConnectionIntegrationTestPlanB, caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); // Wait for ICE to complete, without any tracks being set. - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, caller()->ice_connection_state(), kMaxWaitForFramesMs); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, callee()->ice_connection_state(), kMaxWaitForFramesMs); // Now set the tracks, and expect frames to immediately start flowing. EXPECT_TRUE( @@ -2182,9 +2179,9 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); // Wait for ICE to complete, without any tracks being set. - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, caller()->ice_connection_state(), kMaxWaitForFramesMs); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, callee()->ice_connection_state(), kMaxWaitForFramesMs); // Now set the tracks, and expect frames to immediately start flowing. auto callee_audio_sender = callee()->pc()->GetSenders()[0]; @@ -2279,21 +2276,21 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock, }); PeerConnectionInterface::RTCConfiguration client_1_config; - webrtc::PeerConnectionInterface::IceServer ice_server_1; + PeerConnectionInterface::IceServer ice_server_1; ice_server_1.urls.push_back("turn:88.88.88.0:3478"); ice_server_1.username = "test"; ice_server_1.password = "test"; client_1_config.servers.push_back(ice_server_1); - client_1_config.type = webrtc::PeerConnectionInterface::kRelay; + client_1_config.type = PeerConnectionInterface::kRelay; client_1_config.presume_writable_when_fully_relayed = true; PeerConnectionInterface::RTCConfiguration client_2_config; - webrtc::PeerConnectionInterface::IceServer ice_server_2; + PeerConnectionInterface::IceServer ice_server_2; ice_server_2.urls.push_back("turn:99.99.99.0:3478"); ice_server_2.username = "test"; ice_server_2.password = "test"; client_2_config.servers.push_back(ice_server_2); - client_2_config.type = webrtc::PeerConnectionInterface::kRelay; + client_2_config.type = PeerConnectionInterface::kRelay; client_2_config.presume_writable_when_fully_relayed = true; ASSERT_TRUE( @@ -2326,22 +2323,22 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock, caller()->AddAudioTrack(); // Call getStats, assert there are no candidates. - rtc::scoped_refptr first_report = + rtc::scoped_refptr first_report = caller()->NewGetStats(); ASSERT_TRUE(first_report); auto first_candidate_stats = - first_report->GetStatsOfType(); + first_report->GetStatsOfType(); ASSERT_EQ(first_candidate_stats.size(), 0u); // Create an offer at the caller and set it as remote description on the // callee. caller()->CreateAndSetAndSignalOffer(); // Call getStats again, assert there are candidates now. - rtc::scoped_refptr second_report = + rtc::scoped_refptr second_report = caller()->NewGetStats(); ASSERT_TRUE(second_report); auto second_candidate_stats = - second_report->GetStatsOfType(); + second_report->GetStatsOfType(); ASSERT_NE(second_candidate_stats.size(), 0u); // The fake clock ensures that no time has passed so the cache must have been @@ -2362,17 +2359,17 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock, kDefaultTimeout, FakeClock()); // Call getStats, assert there are no candidates. - rtc::scoped_refptr first_report = + rtc::scoped_refptr first_report = caller()->NewGetStats(); ASSERT_TRUE(first_report); auto first_candidate_stats = - first_report->GetStatsOfType(); + first_report->GetStatsOfType(); ASSERT_EQ(first_candidate_stats.size(), 0u); // Add a "fake" candidate. absl::optional result; caller()->pc()->AddIceCandidate( - absl::WrapUnique(webrtc::CreateIceCandidate( + absl::WrapUnique(CreateIceCandidate( "", 0, "candidate:2214029314 1 udp 2122260223 127.0.0.1 49152 typ host", nullptr)), @@ -2381,11 +2378,11 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock, ASSERT_TRUE(result.value().ok()); // Call getStats again, assert there is a remote candidate now. - rtc::scoped_refptr second_report = + rtc::scoped_refptr second_report = caller()->NewGetStats(); ASSERT_TRUE(second_report); auto second_candidate_stats = - second_report->GetStatsOfType(); + second_report->GetStatsOfType(); ASSERT_EQ(second_candidate_stats.size(), 1u); // The fake clock ensures that no time has passed so the cache must have been @@ -2413,22 +2410,22 @@ TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) { turn_server_2_external_address); PeerConnectionInterface::RTCConfiguration client_1_config; - webrtc::PeerConnectionInterface::IceServer ice_server_1; + PeerConnectionInterface::IceServer ice_server_1; ice_server_1.urls.push_back("turn:88.88.88.0:3478"); ice_server_1.username = "test"; ice_server_1.password = "test"; client_1_config.servers.push_back(ice_server_1); - client_1_config.type = webrtc::PeerConnectionInterface::kRelay; + client_1_config.type = PeerConnectionInterface::kRelay; auto* customizer1 = CreateTurnCustomizer(); client_1_config.turn_customizer = customizer1; PeerConnectionInterface::RTCConfiguration client_2_config; - webrtc::PeerConnectionInterface::IceServer ice_server_2; + PeerConnectionInterface::IceServer ice_server_2; ice_server_2.urls.push_back("turn:99.99.99.0:3478"); ice_server_2.username = "test"; ice_server_2.password = "test"; client_2_config.servers.push_back(ice_server_2); - client_2_config.type = webrtc::PeerConnectionInterface::kRelay; + client_2_config.type = PeerConnectionInterface::kRelay; auto* customizer2 = CreateTurnCustomizer(); client_2_config.turn_customizer = customizer2; @@ -2460,18 +2457,18 @@ TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) { CreateTurnServer(turn_server_internal_address, turn_server_external_address, cricket::PROTO_TCP); - webrtc::PeerConnectionInterface::IceServer ice_server; + PeerConnectionInterface::IceServer ice_server; ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp"); ice_server.username = "test"; ice_server.password = "test"; PeerConnectionInterface::RTCConfiguration client_1_config; client_1_config.servers.push_back(ice_server); - client_1_config.type = webrtc::PeerConnectionInterface::kRelay; + client_1_config.type = PeerConnectionInterface::kRelay; PeerConnectionInterface::RTCConfiguration client_2_config; client_2_config.servers.push_back(ice_server); - client_2_config.type = webrtc::PeerConnectionInterface::kRelay; + client_2_config.type = PeerConnectionInterface::kRelay; ASSERT_TRUE( CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config)); @@ -2482,7 +2479,7 @@ TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) { callee()->AddAudioVideoTracks(); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, callee()->ice_connection_state(), kMaxWaitForFramesMs); MediaExpectations media_expectations; @@ -2506,20 +2503,20 @@ TEST_P(PeerConnectionIntegrationTest, CreateTurnServer(turn_server_internal_address, turn_server_external_address, cricket::PROTO_TLS, "88.88.88.0"); - webrtc::PeerConnectionInterface::IceServer ice_server; + PeerConnectionInterface::IceServer ice_server; ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp"); ice_server.username = "test"; ice_server.password = "test"; PeerConnectionInterface::RTCConfiguration client_1_config; client_1_config.servers.push_back(ice_server); - client_1_config.type = webrtc::PeerConnectionInterface::kRelay; + client_1_config.type = PeerConnectionInterface::kRelay; PeerConnectionInterface::RTCConfiguration client_2_config; client_2_config.servers.push_back(ice_server); // Setting the type to kRelay forces the connection to go through a TURN // server. - client_2_config.type = webrtc::PeerConnectionInterface::kRelay; + client_2_config.type = PeerConnectionInterface::kRelay; // Get a copy to the pointer so we can verify calls later. rtc::TestCertificateVerifier* client_1_cert_verifier = @@ -2530,10 +2527,10 @@ TEST_P(PeerConnectionIntegrationTest, client_2_cert_verifier->verify_certificate_ = true; // Create the dependencies with the test certificate verifier. - webrtc::PeerConnectionDependencies client_1_deps(nullptr); + PeerConnectionDependencies client_1_deps(nullptr); client_1_deps.tls_cert_verifier = std::unique_ptr(client_1_cert_verifier); - webrtc::PeerConnectionDependencies client_2_deps(nullptr); + PeerConnectionDependencies client_2_deps(nullptr); client_2_deps.tls_cert_verifier = std::unique_ptr(client_2_cert_verifier); @@ -2644,7 +2641,7 @@ TEST_P(PeerConnectionIntegrationTest, GetSourcesAudio) { ASSERT_GT(receiver->GetParameters().encodings.size(), 0u); EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc, sources[0].source_id()); - EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type()); + EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type()); } TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) { @@ -2665,7 +2662,7 @@ TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) { ASSERT_GT(sources.size(), 0u); EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc, sources[0].source_id()); - EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type()); + EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type()); } TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesAudio) { @@ -2684,7 +2681,7 @@ TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesAudio) { })(), kDefaultTimeout); ASSERT_GT(sources.size(), 0u); - EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type()); + EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type()); } TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesVideo) { @@ -2703,7 +2700,7 @@ TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesVideo) { })(), kDefaultTimeout); ASSERT_GT(sources.size(), 0u); - EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type()); + EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type()); } // Similar to the above test, except instead of waiting until GetSources() is @@ -2728,7 +2725,7 @@ TEST_P(PeerConnectionIntegrationTest, std::vector sources = receiver->GetSources(); // SSRC history must not be cleared since the reception of the first frame. ASSERT_GT(sources.size(), 0u); - EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type()); + EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type()); } TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetParametersAudio) { @@ -2791,9 +2788,9 @@ TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) { ConnectFakeSignaling(); // Add track using stream 1, do offer/answer. - rtc::scoped_refptr track = + rtc::scoped_refptr track = caller()->CreateLocalAudioTrack(); - rtc::scoped_refptr sender = + rtc::scoped_refptr sender = caller()->AddTrack(track, {"stream_1"}); caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); @@ -2825,8 +2822,8 @@ TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) { .WillByDefault(::testing::Return(true)); EXPECT_CALL(*output, Write(::testing::A())) .Times(::testing::AtLeast(1)); - EXPECT_TRUE(caller()->pc()->StartRtcEventLog( - std::move(output), webrtc::RtcEventLog::kImmediateOutput)); + EXPECT_TRUE(caller()->pc()->StartRtcEventLog(std::move(output), + RtcEventLog::kImmediateOutput)); caller()->AddAudioVideoTracks(); caller()->CreateAndSetAndSignalOffer(); @@ -2900,8 +2897,7 @@ TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) { double GetAudioEnergyStat(PeerConnectionIntegrationWrapper* pc) { auto report = pc->NewGetStats(); - auto inbound_rtps = - report->GetStatsOfType(); + auto inbound_rtps = report->GetStatsOfType(); RTC_CHECK(!inbound_rtps.empty()); auto* inbound_rtp = inbound_rtps[0]; if (!inbound_rtp->total_audio_energy.is_defined()) { @@ -2974,20 +2970,20 @@ TEST_P(PeerConnectionIntegrationTest, ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout); ASSERT_NE(nullptr, caller()->event_log_factory()); ASSERT_NE(nullptr, callee()->event_log_factory()); - webrtc::FakeRtcEventLog* caller_event_log = + FakeRtcEventLog* caller_event_log = caller()->event_log_factory()->last_log_created(); - webrtc::FakeRtcEventLog* callee_event_log = + FakeRtcEventLog* callee_event_log = callee()->event_log_factory()->last_log_created(); ASSERT_NE(nullptr, caller_event_log); ASSERT_NE(nullptr, callee_event_log); - int caller_ice_config_count = caller_event_log->GetEventCount( - webrtc::RtcEvent::Type::IceCandidatePairConfig); - int caller_ice_event_count = caller_event_log->GetEventCount( - webrtc::RtcEvent::Type::IceCandidatePairEvent); - int callee_ice_config_count = callee_event_log->GetEventCount( - webrtc::RtcEvent::Type::IceCandidatePairConfig); - int callee_ice_event_count = callee_event_log->GetEventCount( - webrtc::RtcEvent::Type::IceCandidatePairEvent); + int caller_ice_config_count = + caller_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairConfig); + int caller_ice_event_count = + caller_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairEvent); + int callee_ice_config_count = + callee_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairConfig); + int callee_ice_event_count = + callee_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairEvent); EXPECT_LT(0, caller_ice_config_count); EXPECT_LT(0, caller_ice_event_count); EXPECT_LT(0, callee_ice_config_count); @@ -3001,20 +2997,20 @@ TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) { CreateTurnServer(turn_server_internal_address, turn_server_external_address); - webrtc::PeerConnectionInterface::IceServer ice_server; + PeerConnectionInterface::IceServer ice_server; ice_server.urls.push_back("turn:88.88.88.0:3478"); ice_server.username = "test"; ice_server.password = "test"; PeerConnectionInterface::RTCConfiguration caller_config; caller_config.servers.push_back(ice_server); - caller_config.type = webrtc::PeerConnectionInterface::kRelay; + caller_config.type = PeerConnectionInterface::kRelay; caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY; caller_config.surface_ice_candidates_on_ice_transport_type_changed = true; PeerConnectionInterface::RTCConfiguration callee_config; callee_config.servers.push_back(ice_server); - callee_config.type = webrtc::PeerConnectionInterface::kRelay; + callee_config.type = PeerConnectionInterface::kRelay; callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY; callee_config.surface_ice_candidates_on_ice_transport_type_changed = true; @@ -3031,9 +3027,9 @@ TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) { // kIceGatheringComplete (see // P2PTransportChannel::OnCandidatesAllocationDone), and consequently not // kIceConnectionComplete. - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, caller()->ice_connection_state(), kDefaultTimeout); - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, + EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected, callee()->ice_connection_state(), kDefaultTimeout); // Note that we cannot use the metric // `WebRTC.PeerConnection.CandidatePairType_UDP` in this test since this @@ -3046,7 +3042,7 @@ TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) { // Loosen the caller's candidate filter. caller_config = caller()->pc()->GetConfiguration(); - caller_config.type = webrtc::PeerConnectionInterface::kAll; + caller_config.type = PeerConnectionInterface::kAll; caller()->pc()->SetConfiguration(caller_config); // We should have gathered a new host candidate. EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE, @@ -3054,7 +3050,7 @@ TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) { // Loosen the callee's candidate filter. callee_config = callee()->pc()->GetConfiguration(); - callee_config.type = webrtc::PeerConnectionInterface::kAll; + callee_config.type = PeerConnectionInterface::kAll; callee()->pc()->SetConfiguration(callee_config); EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE, callee()->last_candidate_gathered().type(), kDefaultTimeout); @@ -3084,19 +3080,19 @@ TEST_P(PeerConnectionIntegrationTest, OnIceCandidateError) { CreateTurnServer(turn_server_internal_address, turn_server_external_address); - webrtc::PeerConnectionInterface::IceServer ice_server; + PeerConnectionInterface::IceServer ice_server; ice_server.urls.push_back("turn:88.88.88.0:3478"); ice_server.username = "test"; ice_server.password = "123"; PeerConnectionInterface::RTCConfiguration caller_config; caller_config.servers.push_back(ice_server); - caller_config.type = webrtc::PeerConnectionInterface::kRelay; + caller_config.type = PeerConnectionInterface::kRelay; caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY; PeerConnectionInterface::RTCConfiguration callee_config; callee_config.servers.push_back(ice_server); - callee_config.type = webrtc::PeerConnectionInterface::kRelay; + callee_config.type = PeerConnectionInterface::kRelay; callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY; ASSERT_TRUE( @@ -3115,19 +3111,19 @@ TEST_P(PeerConnectionIntegrationTest, OnIceCandidateError) { } TEST_P(PeerConnectionIntegrationTest, OnIceCandidateErrorWithEmptyAddress) { - webrtc::PeerConnectionInterface::IceServer ice_server; + PeerConnectionInterface::IceServer ice_server; ice_server.urls.push_back("turn:127.0.0.1:3478?transport=tcp"); ice_server.username = "test"; ice_server.password = "test"; PeerConnectionInterface::RTCConfiguration caller_config; caller_config.servers.push_back(ice_server); - caller_config.type = webrtc::PeerConnectionInterface::kRelay; + caller_config.type = PeerConnectionInterface::kRelay; caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY; PeerConnectionInterface::RTCConfiguration callee_config; callee_config.servers.push_back(ice_server); - callee_config.type = webrtc::PeerConnectionInterface::kRelay; + callee_config.type = PeerConnectionInterface::kRelay; callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY; ASSERT_TRUE( @@ -3697,7 +3693,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndRtpSenderVideoEncoderSelector) { CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/true)); ConnectFakeSignaling(); // Add one-directional video, from caller to callee. - rtc::scoped_refptr caller_track = + rtc::scoped_refptr caller_track = caller()->CreateLocalVideoTrack(); auto sender = caller()->AddTrack(caller_track); PeerConnectionInterface::RTCOfferAnswerOptions options; @@ -3722,7 +3718,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndRtpSenderVideoEncoderSelector) { } int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) { - rtc::scoped_refptr report = pc.NewGetStats(); + rtc::scoped_refptr report = pc.NewGetStats(); auto sender_stats = report->GetStatsOfType(); if (sender_stats.size() != 1) { ADD_FAILURE(); @@ -3735,7 +3731,7 @@ int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) { } int NacksSentCount(PeerConnectionIntegrationWrapper& pc) { - rtc::scoped_refptr report = pc.NewGetStats(); + rtc::scoped_refptr report = pc.NewGetStats(); auto receiver_stats = report->GetStatsOfType(); if (receiver_stats.size() != 1) { ADD_FAILURE(); diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc index 2dbc6f552d..c057e55073 100644 --- a/pc/peer_connection_interface_unittest.cc +++ b/pc/peer_connection_interface_unittest.cc @@ -474,8 +474,7 @@ bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) { // Get the ufrags out of an SDP blob. Useful for testing ICE restart // behavior. -std::vector GetUfrags( - const webrtc::SessionDescriptionInterface* desc) { +std::vector GetUfrags(const SessionDescriptionInterface* desc) { std::vector ufrags; for (const cricket::TransportInfo& info : desc->description()->transport_infos()) { @@ -544,21 +543,19 @@ rtc::scoped_refptr CreateStreamCollection( StreamCollection::Create()); for (int i = 0; i < number_of_streams; ++i) { - rtc::scoped_refptr stream( - webrtc::MediaStream::Create(kStreams[i])); + rtc::scoped_refptr stream( + MediaStream::Create(kStreams[i])); for (int j = 0; j < tracks_per_stream; ++j) { // Add a local audio track. - rtc::scoped_refptr audio_track( - webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j], - nullptr)); + rtc::scoped_refptr audio_track( + AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j], nullptr)); stream->AddTrack(audio_track); // Add a local video track. - rtc::scoped_refptr video_track( - webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j], - webrtc::FakeVideoTrackSource::Create(), - rtc::Thread::Current())); + rtc::scoped_refptr video_track(VideoTrack::Create( + kVideoTracks[i * tracks_per_stream + j], + FakeVideoTrackSource::Create(), rtc::Thread::Current())); stream->AddTrack(video_track); } @@ -578,10 +575,10 @@ bool CompareStreamCollections(StreamCollectionInterface* s1, if (s1->at(i)->id() != s2->at(i)->id()) { return false; } - webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks(); - webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks(); - webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks(); - webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks(); + AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks(); + AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks(); + VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks(); + VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks(); if (audio_tracks1.size() != audio_tracks2.size()) { return false; @@ -630,7 +627,7 @@ class MockTrackObserver : public ObserverInterface { // constraints are propagated into the PeerConnection's MediaConfig. These // settings are intended for MediaChannel constructors, but that is not // exercised by these unittest. -class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory { +class PeerConnectionFactoryForTest : public PeerConnectionFactory { public: static rtc::scoped_refptr CreatePeerConnectionFactoryForTest() { @@ -665,7 +662,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { main_(vss_.get()), sdp_semantics_(sdp_semantics) { #ifdef WEBRTC_ANDROID - webrtc::InitializeAndroidObjects(); + InitializeAndroidObjects(); #endif } @@ -673,22 +670,16 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { // Use fake audio capture module since we're only testing the interface // level, and using a real one could make tests flaky when run in parallel. fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); - pc_factory_ = webrtc::CreatePeerConnectionFactory( + pc_factory_ = CreatePeerConnectionFactory( rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(), - rtc::scoped_refptr( - fake_audio_capture_module_), - webrtc::CreateBuiltinAudioEncoderFactory(), - webrtc::CreateBuiltinAudioDecoderFactory(), - std::make_unique>(), - std::make_unique>(), + rtc::scoped_refptr(fake_audio_capture_module_), + CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(), + std::make_unique>(), + std::make_unique>(), nullptr /* audio_mixer */, nullptr /* audio_processing */); ASSERT_TRUE(pc_factory_); } @@ -946,8 +937,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { // Call the standards-compliant GetStats function. bool DoGetRTCStats() { - auto callback = - rtc::make_ref_counted(); + auto callback = rtc::make_ref_counted(); pc_->GetStats(callback.get()); EXPECT_TRUE_WAIT(callback->called(), kTimeout); return callback->called(); @@ -987,14 +977,14 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE(offer->ToString(&sdp)); std::unique_ptr remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_); } void CreateAndSetRemoteOffer(const std::string& sdp) { std::unique_ptr remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_); } @@ -1013,7 +1003,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE(answer->ToString(&sdp)); std::unique_ptr new_answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); EXPECT_TRUE(DoSetLocalDescription(std::move(new_answer))); EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); } @@ -1025,7 +1015,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE(answer->ToString(&sdp)); std::unique_ptr pr_answer( - webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); + CreateSessionDescription(SdpType::kPrAnswer, sdp)); EXPECT_TRUE(DoSetLocalDescription(std::move(pr_answer))); EXPECT_EQ(PeerConnectionInterface::kHaveLocalPrAnswer, observer_.state_); } @@ -1050,7 +1040,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE(offer->ToString(&sdp)); std::unique_ptr new_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_); @@ -1060,7 +1050,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { void CreateAnswerAsRemoteDescription(const std::string& sdp) { std::unique_ptr answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); ASSERT_TRUE(answer); EXPECT_TRUE(DoSetRemoteDescription(std::move(answer))); EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); @@ -1068,12 +1058,12 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) { std::unique_ptr pr_answer( - webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); + CreateSessionDescription(SdpType::kPrAnswer, sdp)); ASSERT_TRUE(pr_answer); EXPECT_TRUE(DoSetRemoteDescription(std::move(pr_answer))); EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_); std::unique_ptr answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); ASSERT_TRUE(answer); EXPECT_TRUE(DoSetRemoteDescription(std::move(answer))); EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); @@ -1117,8 +1107,8 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string mediastream_id = kStreams[0]; - rtc::scoped_refptr stream( - webrtc::MediaStream::Create(mediastream_id)); + rtc::scoped_refptr stream( + MediaStream::Create(mediastream_id)); reference_collection_->AddStream(stream); if (number_of_audio_tracks > 0) { @@ -1142,22 +1132,20 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { } return std::unique_ptr( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp_ms1)); + CreateSessionDescription(SdpType::kOffer, sdp_ms1)); } void AddAudioTrack(const std::string& track_id, MediaStreamInterface* stream) { - rtc::scoped_refptr audio_track( - webrtc::AudioTrack::Create(track_id, nullptr)); + rtc::scoped_refptr audio_track( + AudioTrack::Create(track_id, nullptr)); ASSERT_TRUE(stream->AddTrack(audio_track)); } void AddVideoTrack(const std::string& track_id, MediaStreamInterface* stream) { - rtc::scoped_refptr video_track( - webrtc::VideoTrack::Create(track_id, - webrtc::FakeVideoTrackSource::Create(), - rtc::Thread::Current())); + rtc::scoped_refptr video_track(VideoTrack::Create( + track_id, FakeVideoTrackSource::Create(), rtc::Thread::Current())); ASSERT_TRUE(stream->AddTrack(video_track)); } @@ -1217,7 +1205,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE((*desc)->ToString(&sdp)); std::unique_ptr remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_); } @@ -1230,7 +1218,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE((*desc)->ToString(&sdp)); std::unique_ptr new_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_); @@ -1266,13 +1254,13 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { rtc::SocketServer* socket_server() const { return vss_.get(); } - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; std::unique_ptr vss_; rtc::AutoSocketServerThread main_; rtc::scoped_refptr fake_audio_capture_module_; cricket::FakePortAllocator* port_allocator_ = nullptr; FakeRTCCertificateGenerator* fake_certificate_generator_ = nullptr; - rtc::scoped_refptr pc_factory_; + rtc::scoped_refptr pc_factory_; rtc::scoped_refptr pc_; MockPeerConnectionObserver observer_; rtc::scoped_refptr reference_collection_; @@ -1392,22 +1380,19 @@ TEST_P(PeerConnectionInterfaceTest, config.prune_turn_ports = true; // Create the PC factory and PC with the above config. - rtc::scoped_refptr pc_factory( - webrtc::CreatePeerConnectionFactory( + rtc::scoped_refptr pc_factory( + CreatePeerConnectionFactory( rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(), fake_audio_capture_module_, - webrtc::CreateBuiltinAudioEncoderFactory(), - webrtc::CreateBuiltinAudioDecoderFactory(), - std::make_unique>(), - std::make_unique>(), + CreateBuiltinAudioEncoderFactory(), + CreateBuiltinAudioDecoderFactory(), + std::make_unique>(), + std::make_unique>(), nullptr /* audio_mixer */, nullptr /* audio_processing */)); PeerConnectionDependencies pc_dependencies(&observer_); pc_dependencies.allocator = std::move(port_allocator); @@ -1424,7 +1409,7 @@ TEST_P(PeerConnectionInterfaceTest, EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_TCP); EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); - EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY, + EXPECT_EQ(PRUNE_BASED_ON_PRIORITY, raw_port_allocator->turn_port_prune_policy()); } @@ -1446,8 +1431,7 @@ TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) { TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) { PeerConnectionInterface::RTCConfiguration starting_config; starting_config.sdp_semantics = sdp_semantics_; - starting_config.bundle_policy = - webrtc::PeerConnection::kBundlePolicyMaxBundle; + starting_config.bundle_policy = PeerConnection::kBundlePolicyMaxBundle; CreatePeerConnection(starting_config); PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration(); @@ -1978,7 +1962,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) { RTCConfiguration rtc_config; CreatePeerConnection(rtc_config); - webrtc::DataChannelInit config; + DataChannelInit config; auto channel = pc_->CreateDataChannelOrError("1", &config); EXPECT_TRUE(channel.ok()); EXPECT_TRUE(channel.value()->reliable()); @@ -2010,7 +1994,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWhenClosed) { RTCConfiguration rtc_config; CreatePeerConnection(rtc_config); pc_->Close(); - webrtc::DataChannelInit config; + DataChannelInit config; auto ret = pc_->CreateDataChannelOrError("1", &config); ASSERT_FALSE(ret.ok()); EXPECT_EQ(ret.error().type(), RTCErrorType::INVALID_STATE); @@ -2022,7 +2006,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWithMinusOne) { RTCConfiguration rtc_config; CreatePeerConnection(rtc_config); - webrtc::DataChannelInit config; + DataChannelInit config; config.maxRetransmitTime = -1; config.maxRetransmits = -1; auto channel = pc_->CreateDataChannelOrError("1", &config); @@ -2037,7 +2021,7 @@ TEST_P(PeerConnectionInterfaceTest, CreatePeerConnection(rtc_config); std::string label = "test"; - webrtc::DataChannelInit config; + DataChannelInit config; config.maxRetransmits = 0; config.maxRetransmitTime = 0; @@ -2052,7 +2036,7 @@ TEST_P(PeerConnectionInterfaceTest, RTCConfiguration rtc_config; CreatePeerConnection(rtc_config); - webrtc::DataChannelInit config; + DataChannelInit config; config.id = 1; config.negotiated = true; @@ -2106,7 +2090,7 @@ TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer) std::string sdp; EXPECT_TRUE(pc_->local_description()->ToString(&sdp)); std::unique_ptr answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); ASSERT_TRUE(answer); cricket::ContentInfo* data_info = cricket::GetFirstDataContent(answer->description()); @@ -2125,8 +2109,7 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) { AddAudioTrack("audio_label"); AddVideoTrack("video_label"); std::unique_ptr desc( - webrtc::CreateSessionDescription(SdpType::kOffer, - webrtc::kFireFoxSdpOffer, nullptr)); + CreateSessionDescription(SdpType::kOffer, kFireFoxSdpOffer, nullptr)); EXPECT_TRUE(DoSetSessionDescription(std::move(desc), false)); CreateAnswerAsLocalDescription(); ASSERT_TRUE(pc_->local_description() != nullptr); @@ -2163,8 +2146,7 @@ TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) { EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(), kTimeout); std::unique_ptr desc( - webrtc::CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp, - nullptr)); + CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp, nullptr)); EXPECT_FALSE(DoSetSessionDescription(std::move(desc), /*local=*/false)); } @@ -2177,18 +2159,17 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) { CreateOfferAsLocalDescription(); const char* answer_sdp = (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED - ? webrtc::kAudioSdpPlanB - : webrtc::kAudioSdpUnifiedPlan); + ? kAudioSdpPlanB + : kAudioSdpUnifiedPlan); std::unique_ptr answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr)); + CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr)); EXPECT_TRUE(DoSetSessionDescription(std::move(answer), false)); - const char* reoffer_sdp = - (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED - ? webrtc::kAudioSdpWithUnsupportedCodecsPlanB - : webrtc::kAudioSdpWithUnsupportedCodecsUnifiedPlan); + const char* reoffer_sdp = (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED + ? kAudioSdpWithUnsupportedCodecsPlanB + : kAudioSdpWithUnsupportedCodecsUnifiedPlan); std::unique_ptr updated_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr)); + CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr)); EXPECT_TRUE(DoSetSessionDescription(std::move(updated_offer), false)); CreateAnswerAsLocalDescription(); } @@ -2275,12 +2256,11 @@ TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) { config.prune_turn_ports = false; CreatePeerConnection(config); config = pc_->GetConfiguration(); - EXPECT_EQ(webrtc::NO_PRUNE, port_allocator_->turn_port_prune_policy()); + EXPECT_EQ(NO_PRUNE, port_allocator_->turn_port_prune_policy()); config.prune_turn_ports = true; EXPECT_TRUE(pc_->SetConfiguration(config).ok()); - EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY, - port_allocator_->turn_port_prune_policy()); + EXPECT_EQ(PRUNE_BASED_ON_PRIORITY, port_allocator_->turn_port_prune_policy()); } // Test that the ice check interval can be changed. This does not verify that @@ -2549,12 +2529,12 @@ TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) { std::string sdp; ASSERT_TRUE(pc_->remote_description()->ToString(&sdp)); std::unique_ptr remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_FALSE(DoSetRemoteDescription(std::move(remote_offer))); ASSERT_TRUE(pc_->local_description()->ToString(&sdp)); std::unique_ptr local_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_FALSE(DoSetLocalDescription(std::move(local_offer))); } @@ -2614,10 +2594,10 @@ TEST_F(PeerConnectionInterfaceTestPlanB, reference_collection_.get())); rtc::scoped_refptr audio_track2 = observer_.remote_streams()->at(0)->GetAudioTracks()[1]; - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state()); + EXPECT_EQ(MediaStreamTrackInterface::kLive, audio_track2->state()); rtc::scoped_refptr video_track2 = observer_.remote_streams()->at(0)->GetVideoTracks()[1]; - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state()); + EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track2->state()); // Remove the extra audio and video tracks. std::unique_ptr desc_ms2 = @@ -2631,10 +2611,10 @@ TEST_F(PeerConnectionInterfaceTestPlanB, EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(), reference_collection_.get())); // Track state may be updated asynchronously. - EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded, - audio_track2->state(), kTimeout); - EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded, - video_track2->state(), kTimeout); + EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, audio_track2->state(), + kTimeout); + EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, video_track2->state(), + kTimeout); } // This tests that remote tracks are ended if a local session description is set @@ -2652,7 +2632,7 @@ TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) { rtc::scoped_refptr remote_audio = audio_receiver->track(); - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state()); + EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_audio->state()); rtc::scoped_refptr remote_video = video_receiver->track(); EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_video->state()); @@ -2696,8 +2676,8 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) { remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]); std::unique_ptr local_answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, - GetSdpStringWithStream1(), nullptr)); + CreateSessionDescription(SdpType::kAnswer, GetSdpStringWithStream1(), + nullptr)); cricket::ContentInfo* video_info = local_answer->description()->GetContentByName("video"); video_info->rejected = true; @@ -2986,9 +2966,9 @@ TEST_P(PeerConnectionInterfaceTest, ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); // Grab a copy of the offer before it gets passed into the PC. std::unique_ptr modified_offer = - webrtc::CreateSessionDescription( - webrtc::SdpType::kOffer, offer->session_id(), - offer->session_version(), offer->description()->Clone()); + CreateSessionDescription(SdpType::kOffer, offer->session_id(), + offer->session_version(), + offer->description()->Clone()); EXPECT_TRUE(DoSetLocalDescription(std::move(offer))); auto senders = pc_->GetSenders(); @@ -3044,8 +3024,8 @@ TEST_F(PeerConnectionInterfaceTestPlanB, EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0])); // Add a new MediaStream but with the same tracks as in the first stream. - rtc::scoped_refptr stream_1( - webrtc::MediaStream::Create(kStreams[1])); + rtc::scoped_refptr stream_1( + MediaStream::Create(kStreams[1])); stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]); stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]); pc_->AddStream(stream_1.get()); @@ -3166,9 +3146,9 @@ TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) { EXPECT_TRUE(pc_->SetConfiguration(config).ok()); // Do ICE restart for the first m= section, initiated by remote peer. - std::unique_ptr remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, - GetSdpStringWithStream1(), nullptr)); + std::unique_ptr remote_offer( + CreateSessionDescription(SdpType::kOffer, GetSdpStringWithStream1(), + nullptr)); ASSERT_TRUE(remote_offer); remote_offer->description()->transport_infos()[0].description.ice_ufrag = "modified"; @@ -3214,7 +3194,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set remote pranswer. std::unique_ptr remote_pranswer( - webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); + CreateSessionDescription(SdpType::kPrAnswer, sdp)); SessionDescriptionInterface* remote_pranswer_ptr = remote_pranswer.get(); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_pranswer))); EXPECT_EQ(local_offer_ptr, pc_->pending_local_description()); @@ -3224,7 +3204,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set remote answer. std::unique_ptr remote_answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); SessionDescriptionInterface* remote_answer_ptr = remote_answer.get(); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_answer))); EXPECT_EQ(nullptr, pc_->pending_local_description()); @@ -3234,7 +3214,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set remote offer. std::unique_ptr remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); SessionDescriptionInterface* remote_offer_ptr = remote_offer.get(); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description()); @@ -3244,7 +3224,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set local pranswer. std::unique_ptr local_pranswer( - webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); + CreateSessionDescription(SdpType::kPrAnswer, sdp)); SessionDescriptionInterface* local_pranswer_ptr = local_pranswer.get(); EXPECT_TRUE(DoSetLocalDescription(std::move(local_pranswer))); EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description()); @@ -3254,7 +3234,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set local answer. std::unique_ptr local_answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); SessionDescriptionInterface* local_answer_ptr = local_answer.get(); EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer))); EXPECT_EQ(nullptr, pc_->pending_remote_description()); @@ -3273,9 +3253,8 @@ TEST_P(PeerConnectionInterfaceTest, // The RtcEventLog will be reset when the PeerConnection is closed. pc_->Close(); - EXPECT_FALSE( - pc_->StartRtcEventLog(std::make_unique(), - webrtc::RtcEventLog::kImmediateOutput)); + EXPECT_FALSE(pc_->StartRtcEventLog(std::make_unique(), + RtcEventLog::kImmediateOutput)); pc_->StopRtcEventLog(); } diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc index f08474e5bf..796520f9bc 100644 --- a/pc/peer_connection_media_unittest.cc +++ b/pc/peer_connection_media_unittest.cc @@ -82,9 +82,9 @@ using ::testing::NotNull; using ::testing::Values; cricket::MediaSendChannelInterface* SendChannelInternal( - rtc::scoped_refptr transceiver) { - auto transceiver_with_internal = static_cast>*>( + rtc::scoped_refptr transceiver) { + auto transceiver_with_internal = static_cast< + rtc::RefCountedObject>*>( transceiver.get()); auto transceiver_internal = static_cast(transceiver_with_internal->internal()); @@ -92,9 +92,9 @@ cricket::MediaSendChannelInterface* SendChannelInternal( } cricket::MediaReceiveChannelInterface* ReceiveChannelInternal( - rtc::scoped_refptr transceiver) { - auto transceiver_with_internal = static_cast>*>( + rtc::scoped_refptr transceiver) { + auto transceiver_with_internal = static_cast< + rtc::RefCountedObject>*>( transceiver.get()); auto transceiver_internal = static_cast(transceiver_with_internal->internal()); @@ -102,22 +102,22 @@ cricket::MediaReceiveChannelInterface* ReceiveChannelInternal( } cricket::FakeVideoMediaSendChannel* VideoMediaSendChannel( - rtc::scoped_refptr transceiver) { + rtc::scoped_refptr transceiver) { return static_cast( SendChannelInternal(transceiver)); } cricket::FakeVideoMediaReceiveChannel* VideoMediaReceiveChannel( - rtc::scoped_refptr transceiver) { + rtc::scoped_refptr transceiver) { return static_cast( ReceiveChannelInternal(transceiver)); } cricket::FakeVoiceMediaSendChannel* VoiceMediaSendChannel( - rtc::scoped_refptr transceiver) { + rtc::scoped_refptr transceiver) { return static_cast( SendChannelInternal(transceiver)); } cricket::FakeVoiceMediaReceiveChannel* VoiceMediaReceiveChannel( - rtc::scoped_refptr transceiver) { + rtc::scoped_refptr transceiver) { return static_cast( ReceiveChannelInternal(transceiver)); } @@ -254,7 +254,7 @@ class PeerConnectionMediaBaseTest : public ::testing::Test { return sdp_semantics_ == SdpSemantics::kUnifiedPlan; } - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; std::unique_ptr vss_; rtc::AutoSocketServerThread main_; const SdpSemantics sdp_semantics_; @@ -1495,10 +1495,10 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadDifferentRedundancy) { } template -bool CompareCodecs(const std::vector& capabilities, +bool CompareCodecs(const std::vector& capabilities, const std::vector& codecs) { bool capability_has_rtx = - absl::c_any_of(capabilities, [](const webrtc::RtpCodecCapability& codec) { + absl::c_any_of(capabilities, [](const RtpCodecCapability& codec) { return codec.name == cricket::kRtxCodecName; }); bool codecs_has_rtx = absl::c_any_of(codecs, [](const C& codec) { @@ -1510,16 +1510,16 @@ bool CompareCodecs(const std::vector& capabilities, codecs, std::back_inserter(codecs_no_rtx), [](const C& codec) { return codec.name != cricket::kRtxCodecName; }); - std::vector capabilities_no_rtx; + std::vector capabilities_no_rtx; absl::c_copy_if(capabilities, std::back_inserter(capabilities_no_rtx), - [](const webrtc::RtpCodecCapability& codec) { + [](const RtpCodecCapability& codec) { return codec.name != cricket::kRtxCodecName; }); return capability_has_rtx == codecs_has_rtx && absl::c_equal( capabilities_no_rtx, codecs_no_rtx, - [](const webrtc::RtpCodecCapability& capability, const C& codec) { + [](const RtpCodecCapability& capability, const C& codec) { return codec.MatchesRtpCodec(capability); }); } @@ -1538,9 +1538,9 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto capabilities = caller->pc_factory()->GetRtpSenderCapabilities( cricket::MediaType::MEDIA_TYPE_AUDIO); - std::vector codecs; + std::vector codecs; absl::c_copy_if(capabilities.codecs, std::back_inserter(codecs), - [](const webrtc::RtpCodecCapability& codec) { + [](const RtpCodecCapability& codec) { return codec.name.find("_only_") != std::string::npos; }); @@ -1561,9 +1561,9 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto capabilities = caller->pc_factory()->GetRtpReceiverCapabilities( cricket::MediaType::MEDIA_TYPE_AUDIO); - std::vector codecs; + std::vector codecs; absl::c_copy_if(capabilities.codecs, std::back_inserter(codecs), - [](const webrtc::RtpCodecCapability& codec) { + [](const RtpCodecCapability& codec) { return codec.name.find("_only_") != std::string::npos; }); @@ -1611,7 +1611,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto codecs_only_rtx_red_fec = codecs; auto it = std::remove_if(codecs_only_rtx_red_fec.begin(), codecs_only_rtx_red_fec.end(), - [](const webrtc::RtpCodecCapability& codec) { + [](const RtpCodecCapability& codec) { return !(codec.name == cricket::kRtxCodecName || codec.name == cricket::kRedCodecName || codec.name == cricket::kUlpfecCodecName); @@ -1651,7 +1651,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, caller->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; - std::vector empty_codecs = {}; + std::vector empty_codecs = {}; auto audio_transceiver = caller->pc()->GetTransceivers().front(); @@ -1706,7 +1706,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto codecs_only_rtx_red_fec = codecs; auto it = std::remove_if(codecs_only_rtx_red_fec.begin(), codecs_only_rtx_red_fec.end(), - [](const webrtc::RtpCodecCapability& codec) { + [](const RtpCodecCapability& codec) { return !(codec.name == cricket::kRtxCodecName || codec.name == cricket::kRedCodecName || codec.name == cricket::kUlpfecCodecName); @@ -1747,7 +1747,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) .codecs; - std::vector empty_codecs = {}; + std::vector empty_codecs = {}; auto video_transceiver = caller->pc()->GetTransceivers().front(); @@ -1817,7 +1817,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesVideoWithRtx) { auto video_codecs_vpx_rtx = sender_video_codecs; auto it = std::remove_if(video_codecs_vpx_rtx.begin(), video_codecs_vpx_rtx.end(), - [](const webrtc::RtpCodecCapability& codec) { + [](const RtpCodecCapability& codec) { return codec.name != cricket::kRtxCodecName && codec.name != cricket::kVp8CodecName && codec.name != cricket::kVp9CodecName; @@ -1866,7 +1866,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto video_codecs_vpx = video_codecs; auto it = std::remove_if(video_codecs_vpx.begin(), video_codecs_vpx.end(), - [](const webrtc::RtpCodecCapability& codec) { + [](const RtpCodecCapability& codec) { return codec.name != cricket::kVp8CodecName && codec.name != cricket::kVp9CodecName; }); @@ -1889,7 +1889,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto recv_transceiver = callee->pc()->GetTransceivers().front(); auto video_codecs_vp8_rtx = video_codecs; it = std::remove_if(video_codecs_vp8_rtx.begin(), video_codecs_vp8_rtx.end(), - [](const webrtc::RtpCodecCapability& codec) { + [](const RtpCodecCapability& codec) { bool r = codec.name != cricket::kVp8CodecName && codec.name != cricket::kRtxCodecName; return r; @@ -1936,7 +1936,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto video_codecs_vpx = video_codecs; auto it = std::remove_if(video_codecs_vpx.begin(), video_codecs_vpx.end(), - [](const webrtc::RtpCodecCapability& codec) { + [](const RtpCodecCapability& codec) { return codec.name != cricket::kVp8CodecName && codec.name != cricket::kVp9CodecName; }); diff --git a/pc/peer_connection_rampup_tests.cc b/pc/peer_connection_rampup_tests.cc index 545a1d53d0..0fd3c27f7d 100644 --- a/pc/peer_connection_rampup_tests.cc +++ b/pc/peer_connection_rampup_tests.cc @@ -201,7 +201,7 @@ class PeerConnectionRampUpTest : public ::testing::Test { fake_network_managers_.emplace_back(fake_network_manager); auto observer = std::make_unique(); - webrtc::PeerConnectionDependencies dependencies(observer.get()); + PeerConnectionDependencies dependencies(observer.get()); cricket::BasicPortAllocator* port_allocator = new cricket::BasicPortAllocator(fake_network_manager, firewall_socket_factory_.get()); diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc index b93e5923bb..1a97a4bb44 100644 --- a/pc/peer_connection_rtp_unittest.cc +++ b/pc/peer_connection_rtp_unittest.cc @@ -75,13 +75,13 @@ using ::testing::UnorderedElementsAre; using ::testing::Values; template -class OnSuccessObserver : public webrtc::SetRemoteDescriptionObserverInterface { +class OnSuccessObserver : public SetRemoteDescriptionObserverInterface { public: explicit OnSuccessObserver(MethodFunctor on_success) : on_success_(std::move(on_success)) {} - // webrtc::SetRemoteDescriptionObserverInterface implementation. - void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override { + // SetRemoteDescriptionObserverInterface implementation. + void OnSetRemoteDescriptionComplete(RTCError error) override { RTC_CHECK(error.ok()); on_success_(); } @@ -113,7 +113,7 @@ class PeerConnectionRtpBaseTest : public ::testing::Test { Dav1dDecoderTemplateAdapter>>(), nullptr /* audio_mixer */, nullptr /* audio_processing */)) { - webrtc::metrics::Reset(); + metrics::Reset(); } std::unique_ptr CreatePeerConnection() { @@ -201,7 +201,7 @@ class PeerConnectionRtpTestUnifiedPlan : public PeerConnectionRtpBaseTest { } }; -// These tests cover `webrtc::PeerConnectionObserver` callbacks firing upon +// These tests cover `PeerConnectionObserver` callbacks firing upon // setting the remote description. TEST_P(PeerConnectionRtpTest, AddTrackWithoutStreamFiresOnAddTrack) { @@ -934,8 +934,8 @@ TEST_P(PeerConnectionRtpTest, auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - rtc::scoped_refptr observer = - rtc::make_ref_counted(); + rtc::scoped_refptr observer = + rtc::make_ref_counted(); auto offer = caller->CreateOfferAndSetAsLocal(); callee->pc()->SetRemoteDescription(observer.get(), offer.release()); diff --git a/pc/peer_connection_signaling_unittest.cc b/pc/peer_connection_signaling_unittest.cc index 8ca59fc20c..aeba7efecd 100644 --- a/pc/peer_connection_signaling_unittest.cc +++ b/pc/peer_connection_signaling_unittest.cc @@ -896,8 +896,8 @@ TEST_P(PeerConnectionSignalingTest, UnsupportedContentType) { "m=bogus 9 FOO 0 8\r\n" "c=IN IP4 0.0.0.0\r\n" "a=mid:bogusmid\r\n"; - std::unique_ptr remote_description = - webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr); + std::unique_ptr remote_description = + CreateSessionDescription(SdpType::kOffer, sdp, nullptr); EXPECT_TRUE(caller->SetRemoteDescription(std::move(remote_description))); @@ -977,8 +977,8 @@ TEST_P(PeerConnectionSignalingTest, ReceiveFlexFec) { "a=ssrc-group:FEC-FR 1224551896 1953032773\r\n" "a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n" "a=ssrc:1953032773 cname:/exJcmhSLpyu9FgV\r\n"; - std::unique_ptr remote_description = - webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr); + std::unique_ptr remote_description = + CreateSessionDescription(SdpType::kOffer, sdp, nullptr); EXPECT_TRUE(caller->SetRemoteDescription(std::move(remote_description))); @@ -1033,8 +1033,8 @@ TEST_P(PeerConnectionSignalingTest, ReceiveFlexFecReoffer) { "a=ssrc-group:FEC-FR 1224551896 1953032773\r\n" "a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n" "a=ssrc:1953032773 cname:/exJcmhSLpyu9FgV\r\n"; - std::unique_ptr remote_description = - webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr); + std::unique_ptr remote_description = + CreateSessionDescription(SdpType::kOffer, sdp, nullptr); EXPECT_TRUE(caller->SetRemoteDescription(std::move(remote_description))); @@ -1104,8 +1104,8 @@ TEST_P(PeerConnectionSignalingTest, MidAttributeMaxLength) { "a=rtcp-fb:102 nack\r\n" "a=rtcp-fb:102 nack pli\r\n" "a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n"; - std::unique_ptr remote_description = - webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr); + std::unique_ptr remote_description = + CreateSessionDescription(SdpType::kOffer, sdp, nullptr); EXPECT_FALSE(caller->SetRemoteDescription(std::move(remote_description))); } @@ -1339,8 +1339,8 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest, RtxReofferApt) { "a=rtcp-fb:102 nack\r\n" "a=rtcp-fb:102 nack pli\r\n" "a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n"; - std::unique_ptr remote_description = - webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr); + std::unique_ptr remote_description = + CreateSessionDescription(SdpType::kOffer, sdp, nullptr); EXPECT_TRUE(callee->SetRemoteDescription(std::move(remote_description))); diff --git a/pc/peer_connection_simulcast_unittest.cc b/pc/peer_connection_simulcast_unittest.cc index 6b6a96c473..bffb5d9e9f 100644 --- a/pc/peer_connection_simulcast_unittest.cc +++ b/pc/peer_connection_simulcast_unittest.cc @@ -220,7 +220,7 @@ class PeerConnectionSimulcastMetricsTests : public PeerConnectionSimulcastTests, public ::testing::WithParamInterface { protected: - PeerConnectionSimulcastMetricsTests() { webrtc::metrics::Reset(); } + PeerConnectionSimulcastMetricsTests() { metrics::Reset(); } }; #endif diff --git a/pc/peer_connection_svc_integrationtest.cc b/pc/peer_connection_svc_integrationtest.cc index 672f3eef99..32ca451866 100644 --- a/pc/peer_connection_svc_integrationtest.cc +++ b/pc/peer_connection_svc_integrationtest.cc @@ -37,14 +37,13 @@ class PeerConnectionSVCIntegrationTest : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {} RTCError SetCodecPreferences( - rtc::scoped_refptr transceiver, + rtc::scoped_refptr transceiver, absl::string_view codec_name) { - webrtc::RtpCapabilities capabilities = + RtpCapabilities capabilities = caller()->pc_factory()->GetRtpSenderCapabilities( cricket::MEDIA_TYPE_VIDEO); std::vector codecs; - for (const webrtc::RtpCodecCapability& codec_capability : - capabilities.codecs) { + for (const RtpCodecCapability& codec_capability : capabilities.codecs) { if (codec_capability.name == codec_name) codecs.push_back(codec_capability); } @@ -55,8 +54,8 @@ class PeerConnectionSVCIntegrationTest TEST_F(PeerConnectionSVCIntegrationTest, AddTransceiverAcceptsL1T1) { ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; encoding_parameters.scalability_mode = "L1T1"; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = @@ -67,8 +66,8 @@ TEST_F(PeerConnectionSVCIntegrationTest, AddTransceiverAcceptsL1T1) { TEST_F(PeerConnectionSVCIntegrationTest, AddTransceiverAcceptsL3T3) { ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; encoding_parameters.scalability_mode = "L3T3"; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = @@ -80,33 +79,32 @@ TEST_F(PeerConnectionSVCIntegrationTest, AddTransceiverRejectsUnknownScalabilityMode) { ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; encoding_parameters.scalability_mode = "FOOBAR"; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init); EXPECT_FALSE(transceiver_or_error.ok()); EXPECT_EQ(transceiver_or_error.error().type(), - webrtc::RTCErrorType::UNSUPPORTED_OPERATION); + RTCErrorType::UNSUPPORTED_OPERATION); } TEST_F(PeerConnectionSVCIntegrationTest, SetParametersAcceptsL1T3WithVP8) { ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpCapabilities capabilities = + RtpCapabilities capabilities = caller()->pc_factory()->GetRtpSenderCapabilities( cricket::MEDIA_TYPE_VIDEO); std::vector vp8_codec; - for (const webrtc::RtpCodecCapability& codec_capability : - capabilities.codecs) { + for (const RtpCodecCapability& codec_capability : capabilities.codecs) { if (codec_capability.name == cricket::kVp8CodecName) vp8_codec.push_back(codec_capability); } - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init); @@ -114,7 +112,7 @@ TEST_F(PeerConnectionSVCIntegrationTest, SetParametersAcceptsL1T3WithVP8) { auto transceiver = transceiver_or_error.MoveValue(); EXPECT_TRUE(transceiver->SetCodecPreferences(vp8_codec).ok()); - webrtc::RtpParameters parameters = transceiver->sender()->GetParameters(); + RtpParameters parameters = transceiver->sender()->GetParameters(); ASSERT_EQ(parameters.encodings.size(), 1u); parameters.encodings[0].scalability_mode = "L1T3"; auto result = transceiver->sender()->SetParameters(parameters); @@ -125,8 +123,8 @@ TEST_F(PeerConnectionSVCIntegrationTest, SetParametersRejectsL3T3WithVP8) { ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init); @@ -134,12 +132,12 @@ TEST_F(PeerConnectionSVCIntegrationTest, SetParametersRejectsL3T3WithVP8) { auto transceiver = transceiver_or_error.MoveValue(); EXPECT_TRUE(SetCodecPreferences(transceiver, cricket::kVp8CodecName).ok()); - webrtc::RtpParameters parameters = transceiver->sender()->GetParameters(); + RtpParameters parameters = transceiver->sender()->GetParameters(); ASSERT_EQ(parameters.encodings.size(), 1u); parameters.encodings[0].scalability_mode = "L3T3"; auto result = transceiver->sender()->SetParameters(parameters); EXPECT_FALSE(result.ok()); - EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION); + EXPECT_EQ(result.type(), RTCErrorType::INVALID_MODIFICATION); } TEST_F(PeerConnectionSVCIntegrationTest, @@ -147,8 +145,8 @@ TEST_F(PeerConnectionSVCIntegrationTest, ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init); @@ -159,7 +157,7 @@ TEST_F(PeerConnectionSVCIntegrationTest, caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - webrtc::RtpParameters parameters = transceiver->sender()->GetParameters(); + RtpParameters parameters = transceiver->sender()->GetParameters(); ASSERT_EQ(parameters.encodings.size(), 1u); parameters.encodings[0].scalability_mode = "L1T3"; auto result = transceiver->sender()->SetParameters(parameters); @@ -171,8 +169,8 @@ TEST_F(PeerConnectionSVCIntegrationTest, ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init); @@ -183,7 +181,7 @@ TEST_F(PeerConnectionSVCIntegrationTest, caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - webrtc::RtpParameters parameters = transceiver->sender()->GetParameters(); + RtpParameters parameters = transceiver->sender()->GetParameters(); ASSERT_EQ(parameters.encodings.size(), 1u); parameters.encodings[0].scalability_mode = "L3T3"; auto result = transceiver->sender()->SetParameters(parameters); @@ -195,8 +193,8 @@ TEST_F(PeerConnectionSVCIntegrationTest, ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init); @@ -207,12 +205,12 @@ TEST_F(PeerConnectionSVCIntegrationTest, caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - webrtc::RtpParameters parameters = transceiver->sender()->GetParameters(); + RtpParameters parameters = transceiver->sender()->GetParameters(); ASSERT_EQ(parameters.encodings.size(), 1u); parameters.encodings[0].scalability_mode = "L3T3"; auto result = transceiver->sender()->SetParameters(parameters); EXPECT_FALSE(result.ok()); - EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION); + EXPECT_EQ(result.type(), RTCErrorType::INVALID_MODIFICATION); } TEST_F(PeerConnectionSVCIntegrationTest, @@ -220,8 +218,8 @@ TEST_F(PeerConnectionSVCIntegrationTest, ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init); @@ -232,27 +230,27 @@ TEST_F(PeerConnectionSVCIntegrationTest, caller()->CreateAndSetAndSignalOffer(); ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); - webrtc::RtpParameters parameters = transceiver->sender()->GetParameters(); + RtpParameters parameters = transceiver->sender()->GetParameters(); ASSERT_EQ(parameters.encodings.size(), 1u); parameters.encodings[0].scalability_mode = "FOOBAR"; auto result = transceiver->sender()->SetParameters(parameters); EXPECT_FALSE(result.ok()); - EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION); + EXPECT_EQ(result.type(), RTCErrorType::INVALID_MODIFICATION); } TEST_F(PeerConnectionSVCIntegrationTest, FallbackToL1Tx) { ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - webrtc::RtpTransceiverInit init; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + RtpEncodingParameters encoding_parameters; init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init); ASSERT_TRUE(transceiver_or_error.ok()); auto caller_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpCapabilities capabilities = + RtpCapabilities capabilities = caller()->pc_factory()->GetRtpSenderCapabilities( cricket::MEDIA_TYPE_VIDEO); std::vector send_codecs = capabilities.codecs; @@ -267,8 +265,7 @@ TEST_F(PeerConnectionSVCIntegrationTest, FallbackToL1Tx) { caller_transceiver->SetCodecPreferences(send_codecs); // L3T3 should be supported by VP9 - webrtc::RtpParameters parameters = - caller_transceiver->sender()->GetParameters(); + RtpParameters parameters = caller_transceiver->sender()->GetParameters(); ASSERT_EQ(parameters.encodings.size(), 1u); parameters.encodings[0].scalability_mode = "L3T3"; auto result = caller_transceiver->sender()->SetParameters(parameters); diff --git a/pc/peer_connection_wrapper.cc b/pc/peer_connection_wrapper.cc index 44f4256b10..557d0c8422 100644 --- a/pc/peer_connection_wrapper.cc +++ b/pc/peer_connection_wrapper.cc @@ -339,8 +339,7 @@ bool PeerConnectionWrapper::IsIceConnected() { return observer()->ice_connected_; } -rtc::scoped_refptr -PeerConnectionWrapper::GetStats() { +rtc::scoped_refptr PeerConnectionWrapper::GetStats() { auto callback = rtc::make_ref_counted(); pc()->GetStats(callback.get()); EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout); diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc index 0797ba2a76..2bac176aac 100644 --- a/pc/rtc_stats_collector.cc +++ b/pc/rtc_stats_collector.cc @@ -336,7 +336,7 @@ const char* QualityLimitationReasonToRTCQualityLimitationReason( std::map QualityLimitationDurationToRTCQualityLimitationDuration( - std::map durations_ms) { + std::map durations_ms) { std::map result; // The internal duration is defined in milliseconds while the spec defines // the value in seconds: @@ -513,7 +513,7 @@ std::unique_ptr CreateInboundAudioStreamStats( std::unique_ptr CreateAudioPlayoutStats( const AudioDeviceModule::Stats& audio_device_stats, - webrtc::Timestamp timestamp) { + Timestamp timestamp) { auto stats = std::make_unique( /*id=*/kAudioPlayoutSingletonId, timestamp); stats->synthesized_samples_duration = diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h index e94d23944c..4c68e77086 100644 --- a/pc/rtc_stats_collector.h +++ b/pc/rtc_stats_collector.h @@ -317,7 +317,7 @@ class RTCStatsCollector : public rtc::RefCountInterface { uint32_t data_channels_closed; // Identifies channels that have been opened, whose internal id is stored in // the set until they have been fully closed. - webrtc::flat_set opened_data_channels; + flat_set opened_data_channels; }; InternalRecord internal_record_; }; diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc index 37821ac829..055be6fe99 100644 --- a/pc/rtc_stats_collector_unittest.cc +++ b/pc/rtc_stats_collector_unittest.cc @@ -263,9 +263,9 @@ class FakeAudioTrackForStats : public MediaStreamTrack { std::string kind() const override { return MediaStreamTrackInterface::kAudioKind; } - webrtc::AudioSourceInterface* GetSource() const override { return nullptr; } - void AddSink(webrtc::AudioTrackSinkInterface* sink) override {} - void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override {} + AudioSourceInterface* GetSource() const override { return nullptr; } + void AddSink(AudioTrackSinkInterface* sink) override {} + void RemoveSink(AudioTrackSinkInterface* sink) override {} bool GetSignalLevel(int* level) override { return false; } rtc::scoped_refptr GetAudioProcessor() override { return processor_; @@ -2030,7 +2030,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) { EXPECT_TRUE(report->Get(*expected_pair.transport_id)); // Set bandwidth and "GetStats" again. - webrtc::Call::Stats call_stats; + Call::Stats call_stats; const int kSendBandwidth = 888; call_stats.send_bandwidth_bps = kSendBandwidth; const int kRecvBandwidth = 999; @@ -2339,12 +2339,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) { video_media_info.receivers[0].key_frames_decoded = 3; video_media_info.receivers[0].frames_dropped = 13; video_media_info.receivers[0].qp_sum = absl::nullopt; - video_media_info.receivers[0].total_decode_time = - webrtc::TimeDelta::Seconds(9); - video_media_info.receivers[0].total_processing_delay = - webrtc::TimeDelta::Millis(600); - video_media_info.receivers[0].total_assembly_time = - webrtc::TimeDelta::Millis(500); + video_media_info.receivers[0].total_decode_time = TimeDelta::Seconds(9); + video_media_info.receivers[0].total_processing_delay = TimeDelta::Millis(600); + video_media_info.receivers[0].total_assembly_time = TimeDelta::Millis(500); video_media_info.receivers[0].frames_assembled_from_multiple_packets = 23; video_media_info.receivers[0].total_inter_frame_delay = 0.123; video_media_info.receivers[0].total_squared_inter_frame_delay = 0.00456; @@ -2617,12 +2614,12 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) { video_media_info.senders[0].key_frames_encoded = 3; video_media_info.senders[0].total_encode_time_ms = 9000; video_media_info.senders[0].total_encoded_bytes_target = 1234; - video_media_info.senders[0].total_packet_send_delay = - webrtc::TimeDelta::Seconds(10); + video_media_info.senders[0].total_packet_send_delay = TimeDelta::Seconds(10); video_media_info.senders[0].quality_limitation_reason = QualityLimitationReason::kBandwidth; - video_media_info.senders[0].quality_limitation_durations_ms - [webrtc::QualityLimitationReason::kBandwidth] = 300; + video_media_info.senders[0] + .quality_limitation_durations_ms[QualityLimitationReason::kBandwidth] = + 300; video_media_info.senders[0].quality_limitation_resolution_changes = 56u; video_media_info.senders[0].qp_sum = absl::nullopt; video_media_info.senders[0].content_type = VideoContentType::UNSPECIFIED; diff --git a/pc/rtc_stats_traversal_unittest.cc b/pc/rtc_stats_traversal_unittest.cc index 72ad255564..8205ebedc0 100644 --- a/pc/rtc_stats_traversal_unittest.cc +++ b/pc/rtc_stats_traversal_unittest.cc @@ -47,7 +47,7 @@ class RTCStatsTraversalTest : public ::testing::Test { for (const RTCStats* start_node : start_nodes) { start_ids.push_back(start_node->id()); } - result_ = webrtc::TakeReferencedStats(initial_report_, start_ids); + result_ = ::webrtc::TakeReferencedStats(initial_report_, start_ids); } void EXPECT_VISITED(const RTCStats* stats) { diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc index cdae1595b3..b0c32eff85 100644 --- a/pc/rtp_sender.cc +++ b/pc/rtp_sender.cc @@ -115,13 +115,13 @@ class SignalingThreadCallback { if (!signaling_thread_->IsCurrent()) { signaling_thread_->PostTask( [callback = std::move(callback_), error]() mutable { - webrtc::InvokeSetParametersCallback(callback, error); + InvokeSetParametersCallback(callback, error); }); callback_ = nullptr; return; } - webrtc::InvokeSetParametersCallback(callback_, error); + InvokeSetParametersCallback(callback_, error); callback_ = nullptr; } @@ -243,7 +243,7 @@ void RtpSenderBase::SetParametersInternal(const RtpParameters& parameters, "Attempted to set an unimplemented parameter of RtpParameters."); RTC_LOG(LS_ERROR) << error.message() << " (" << ::webrtc::ToString(error.type()) << ")"; - webrtc::InvokeSetParametersCallback(callback, error); + InvokeSetParametersCallback(callback, error); return; } if (!media_channel_ || !ssrc_) { @@ -252,7 +252,7 @@ void RtpSenderBase::SetParametersInternal(const RtpParameters& parameters, if (result.ok()) { init_parameters_ = parameters; } - webrtc::InvokeSetParametersCallback(callback, result); + InvokeSetParametersCallback(callback, result); return; } auto task = [&, callback = std::move(callback), @@ -268,13 +268,13 @@ void RtpSenderBase::SetParametersInternal(const RtpParameters& parameters, RTCError result = cricket::CheckRtpParametersInvalidModificationAndValues( old_parameters, rtp_parameters); if (!result.ok()) { - webrtc::InvokeSetParametersCallback(callback, result); + InvokeSetParametersCallback(callback, result); return; } result = CheckCodecParameters(rtp_parameters); if (!result.ok()) { - webrtc::InvokeSetParametersCallback(callback, result); + InvokeSetParametersCallback(callback, result); return; } @@ -389,7 +389,7 @@ void RtpSenderBase::SetParametersAsync(const RtpParameters& parameters, TRACE_EVENT0("webrtc", "RtpSenderBase::SetParametersAsync"); RTCError result = CheckSetParameters(parameters); if (!result.ok()) { - webrtc::InvokeSetParametersCallback(callback, result); + InvokeSetParametersCallback(callback, result); return; } @@ -399,7 +399,7 @@ void RtpSenderBase::SetParametersAsync(const RtpParameters& parameters, signaling_thread_, [this, callback = std::move(callback)](RTCError error) mutable { last_transaction_id_.reset(); - webrtc::InvokeSetParametersCallback(callback, error); + InvokeSetParametersCallback(callback, error); }), false); } diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h index d29c3760e6..26adceb089 100644 --- a/pc/rtp_sender.h +++ b/pc/rtp_sender.h @@ -87,7 +87,7 @@ class RtpSenderInternal : public RtpSenderInterface { // Additional checks that are specific to the current codec settings virtual RTCError CheckCodecParameters(const RtpParameters& parameters) { - return webrtc::RTCError::OK(); + return RTCError::OK(); } // Returns an ID that changes every time SetTrack() is called, but diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc index 3092e53c2d..4387aedf53 100644 --- a/pc/rtp_sender_receiver_unittest.cc +++ b/pc/rtp_sender_receiver_unittest.cc @@ -105,7 +105,7 @@ class RtpSenderReceiverTest : network_thread_(rtc::Thread::Current()), worker_thread_(rtc::Thread::Current()), video_bitrate_allocator_factory_( - webrtc::CreateBuiltinVideoBitrateAllocatorFactory()), + CreateBuiltinVideoBitrateAllocatorFactory()), // Create fake media engine/etc. so we can create channels to use to // test RtpSenders/RtpReceivers. media_engine_(std::make_unique()), @@ -119,16 +119,16 @@ class RtpSenderReceiverTest // Fake media channels are owned by the media engine. voice_media_send_channel_ = media_engine_->voice().CreateSendChannel( &fake_call_, cricket::MediaConfig(), cricket::AudioOptions(), - webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + CryptoOptions(), AudioCodecPairId::Create()); video_media_send_channel_ = media_engine_->video().CreateSendChannel( &fake_call_, cricket::MediaConfig(), cricket::VideoOptions(), - webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get()); + CryptoOptions(), video_bitrate_allocator_factory_.get()); voice_media_receive_channel_ = media_engine_->voice().CreateReceiveChannel( &fake_call_, cricket::MediaConfig(), cricket::AudioOptions(), - webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + CryptoOptions(), AudioCodecPairId::Create()); video_media_receive_channel_ = media_engine_->video().CreateReceiveChannel( &fake_call_, cricket::MediaConfig(), cricket::VideoOptions(), - webrtc::CryptoOptions()); + CryptoOptions()); // Create streams for predefined SSRCs. Streams need to exist in order // for the senders and receievers to apply parameters to them. @@ -162,8 +162,8 @@ class RtpSenderReceiverTest audio_track_ = nullptr; } - std::unique_ptr CreateDtlsSrtpTransport() { - auto dtls_srtp_transport = std::make_unique( + std::unique_ptr CreateDtlsSrtpTransport() { + auto dtls_srtp_transport = std::make_unique( /*rtcp_mux_required=*/true, field_trials_); dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(), /*rtcp_dtls_transport=*/nullptr); @@ -515,12 +515,12 @@ class RtpSenderReceiverTest test::RunLoop run_loop_; rtc::Thread* const network_thread_; rtc::Thread* const worker_thread_; - webrtc::RtcEventLogNull event_log_; + RtcEventLogNull event_log_; // The `rtp_dtls_transport_` and `rtp_transport_` should be destroyed after // the `channel_manager`. std::unique_ptr rtp_dtls_transport_; - std::unique_ptr rtp_transport_; - std::unique_ptr + std::unique_ptr rtp_transport_; + std::unique_ptr video_bitrate_allocator_factory_; std::unique_ptr media_engine_; rtc::UniqueRandomIdGenerator ssrc_generator_; @@ -540,7 +540,7 @@ class RtpSenderReceiverTest rtc::scoped_refptr local_stream_; rtc::scoped_refptr video_track_; rtc::scoped_refptr audio_track_; - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; }; // Test that `voice_channel_` is updated when an audio track is associated @@ -651,15 +651,13 @@ TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) { TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) { CreateVideoRtpReceiver(); - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state()); - EXPECT_EQ(webrtc::MediaSourceInterface::kLive, - video_track_->GetSource()->state()); + EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track_->state()); + EXPECT_EQ(MediaSourceInterface::kLive, video_track_->GetSource()->state()); DestroyVideoRtpReceiver(); - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state()); - EXPECT_EQ(webrtc::MediaSourceInterface::kEnded, - video_track_->GetSource()->state()); + EXPECT_EQ(MediaStreamTrackInterface::kEnded, video_track_->state()); + EXPECT_EQ(MediaSourceInterface::kEnded, video_track_->GetSource()->state()); DestroyVideoRtpReceiver(); } @@ -888,9 +886,9 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParametersAsync) { RtpParameters params = audio_rtp_sender_->GetParameters(); EXPECT_EQ(1u, params.encodings.size()); - absl::optional result; + absl::optional result; audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -918,13 +916,13 @@ TEST_F(RtpSenderReceiverTest, audio_rtp_sender_ = AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr); - absl::optional result; + absl::optional result; RtpParameters params = audio_rtp_sender_->GetParameters(); ASSERT_EQ(1u, params.encodings.size()); params.encodings[0].max_bitrate_bps = 90000; audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -932,7 +930,7 @@ TEST_F(RtpSenderReceiverTest, EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000); audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -1016,13 +1014,13 @@ TEST_F(RtpSenderReceiverTest, RtpParameters params = audio_rtp_sender_->GetParameters(); EXPECT_EQ(1u, params.encodings.size()); - absl::optional result; + absl::optional result; audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type()); @@ -1081,7 +1079,7 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) { CreateAudioRtpSender(); EXPECT_EQ(-1, voice_media_send_channel()->max_bps()); - webrtc::RtpParameters params = audio_rtp_sender_->GetParameters(); + RtpParameters params = audio_rtp_sender_->GetParameters(); EXPECT_EQ(1U, params.encodings.size()); EXPECT_FALSE(params.encodings[0].max_bitrate_bps); params.encodings[0].max_bitrate_bps = 1000; @@ -1106,10 +1104,9 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) { TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) { CreateAudioRtpSender(); - webrtc::RtpParameters params = audio_rtp_sender_->GetParameters(); + RtpParameters params = audio_rtp_sender_->GetParameters(); EXPECT_EQ(1U, params.encodings.size()); - EXPECT_EQ(webrtc::kDefaultBitratePriority, - params.encodings[0].bitrate_priority); + EXPECT_EQ(kDefaultBitratePriority, params.encodings[0].bitrate_priority); double new_bitrate_priority = 2.0; params.encodings[0].bitrate_priority = new_bitrate_priority; EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok()); @@ -1140,9 +1137,9 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParametersAsync) { RtpParameters params = video_rtp_sender_->GetParameters(); EXPECT_EQ(1u, params.encodings.size()); - absl::optional result; + absl::optional result; video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -1170,19 +1167,19 @@ TEST_F(RtpSenderReceiverTest, video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr); - absl::optional result; + absl::optional result; RtpParameters params = video_rtp_sender_->GetParameters(); ASSERT_EQ(1u, params.encodings.size()); params.encodings[0].max_bitrate_bps = 90000; video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); params = video_rtp_sender_->GetParameters(); EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000); video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -1350,13 +1347,13 @@ TEST_F(RtpSenderReceiverTest, RtpParameters params = video_rtp_sender_->GetParameters(); EXPECT_EQ(1u, params.encodings.size()); - absl::optional result; + absl::optional result; video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type()); @@ -1453,7 +1450,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderDetectInvalidNumTemporalLayers) { CreateVideoRtpSender(); RtpParameters params = video_rtp_sender_->GetParameters(); - params.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1; + params.encodings[0].num_temporal_layers = kMaxTemporalStreams + 1; RTCError result = video_rtp_sender_->SetParameters(params); EXPECT_EQ(RTCErrorType::INVALID_RANGE, result.type()); @@ -1536,7 +1533,7 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrate) { CreateVideoRtpSender(); EXPECT_EQ(-1, video_media_send_channel()->max_bps()); - webrtc::RtpParameters params = video_rtp_sender_->GetParameters(); + RtpParameters params = video_rtp_sender_->GetParameters(); EXPECT_EQ(1U, params.encodings.size()); EXPECT_FALSE(params.encodings[0].min_bitrate_bps); EXPECT_FALSE(params.encodings[0].max_bitrate_bps); @@ -1589,10 +1586,9 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrateSimulcast) { TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) { CreateVideoRtpSender(); - webrtc::RtpParameters params = video_rtp_sender_->GetParameters(); + RtpParameters params = video_rtp_sender_->GetParameters(); EXPECT_EQ(1U, params.encodings.size()); - EXPECT_EQ(webrtc::kDefaultBitratePriority, - params.encodings[0].bitrate_priority); + EXPECT_EQ(kDefaultBitratePriority, params.encodings[0].bitrate_priority); double new_bitrate_priority = 2.0; params.encodings[0].bitrate_priority = new_bitrate_priority; EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok()); diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc index 815ec9dece..ca626cc94b 100644 --- a/pc/rtp_transceiver.cc +++ b/pc/rtp_transceiver.cc @@ -542,7 +542,7 @@ bool RtpTransceiver::stopping() const { RtpTransceiverDirection RtpTransceiver::direction() const { if (unified_plan_ && stopping()) - return webrtc::RtpTransceiverDirection::kStopped; + return RtpTransceiverDirection::kStopped; return direction_; } @@ -570,7 +570,7 @@ RTCError RtpTransceiver::SetDirectionWithError( absl::optional RtpTransceiver::current_direction() const { if (unified_plan_ && stopped()) - return webrtc::RtpTransceiverDirection::kStopped; + return RtpTransceiverDirection::kStopped; return current_direction_; } @@ -604,7 +604,7 @@ void RtpTransceiver::StopSendingAndReceiving() { }); stopping_ = true; - direction_ = webrtc::RtpTransceiverDirection::kInactive; + direction_ = RtpTransceiverDirection::kInactive; } RTCError RtpTransceiver::StopStandard() { diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h index deda5d7d61..88febb9429 100644 --- a/pc/rtp_transceiver.h +++ b/pc/rtp_transceiver.h @@ -358,20 +358,18 @@ PROXY_CONSTMETHOD0(rtc::scoped_refptr, receiver) PROXY_CONSTMETHOD0(bool, stopped) PROXY_CONSTMETHOD0(bool, stopping) PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction) -PROXY_METHOD1(webrtc::RTCError, SetDirectionWithError, RtpTransceiverDirection) +PROXY_METHOD1(RTCError, SetDirectionWithError, RtpTransceiverDirection) PROXY_CONSTMETHOD0(absl::optional, current_direction) PROXY_CONSTMETHOD0(absl::optional, fired_direction) -PROXY_METHOD0(webrtc::RTCError, StopStandard) +PROXY_METHOD0(RTCError, StopStandard) PROXY_METHOD0(void, StopInternal) -PROXY_METHOD1(webrtc::RTCError, - SetCodecPreferences, - rtc::ArrayView) +PROXY_METHOD1(RTCError, SetCodecPreferences, rtc::ArrayView) PROXY_CONSTMETHOD0(std::vector, codec_preferences) PROXY_CONSTMETHOD0(std::vector, GetHeaderExtensionsToNegotiate) PROXY_CONSTMETHOD0(std::vector, GetNegotiatedHeaderExtensions) -PROXY_METHOD1(webrtc::RTCError, +PROXY_METHOD1(RTCError, SetHeaderExtensionsToNegotiate, rtc::ArrayView) END_PROXY_MAP(RtpTransceiver) diff --git a/pc/rtp_transceiver_unittest.cc b/pc/rtp_transceiver_unittest.cc index 63e06bea62..bd711f1a95 100644 --- a/pc/rtp_transceiver_unittest.cc +++ b/pc/rtp_transceiver_unittest.cc @@ -420,8 +420,8 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, ReturnsNegotiatedHdrExts) { EXPECT_CALL(*mock_channel, mid()).WillRepeatedly(ReturnRef(content_name)); EXPECT_CALL(*mock_channel, SetRtpTransport(_)).WillRepeatedly(Return(true)); - cricket::RtpHeaderExtensions extensions = {webrtc::RtpExtension("uri1", 1), - webrtc::RtpExtension("uri2", 2)}; + cricket::RtpHeaderExtensions extensions = {RtpExtension("uri1", 1), + RtpExtension("uri2", 2)}; cricket::AudioContentDescription description; description.set_rtp_header_extensions(extensions); transceiver_->OnNegotiationUpdate(SdpType::kAnswer, &description); @@ -449,8 +449,8 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, EXPECT_CALL(*sender_.get(), SetTransceiverAsStopped()); EXPECT_CALL(*sender_.get(), Stop()); - cricket::RtpHeaderExtensions extensions = {webrtc::RtpExtension("uri1", 1), - webrtc::RtpExtension("uri2", 2)}; + cricket::RtpHeaderExtensions extensions = {RtpExtension("uri1", 1), + RtpExtension("uri2", 2)}; cricket::AudioContentDescription description; description.set_rtp_header_extensions(extensions); transceiver_->OnNegotiationUpdate(SdpType::kAnswer, &description); @@ -464,8 +464,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, RtpTransceiverDirection::kStopped), Field(&RtpHeaderExtensionCapability::direction, RtpTransceiverDirection::kStopped))); - extensions = {webrtc::RtpExtension("uri3", 4), - webrtc::RtpExtension("uri5", 6)}; + extensions = {RtpExtension("uri3", 4), RtpExtension("uri5", 6)}; description.set_rtp_header_extensions(extensions); transceiver_->OnNegotiationUpdate(SdpType::kAnswer, &description); diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc index 2ffb53f477..7cf9fe0ace 100644 --- a/pc/rtp_transport.cc +++ b/pc/rtp_transport.cc @@ -186,10 +186,10 @@ flat_set RtpTransport::GetSsrcsForSink(RtpPacketSinkInterface* sink) { void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us) { - webrtc::RtpPacketReceived parsed_packet( - &header_extension_map_, packet_time_us == -1 - ? Timestamp::MinusInfinity() - : Timestamp::Micros(packet_time_us)); + RtpPacketReceived parsed_packet(&header_extension_map_, + packet_time_us == -1 + ? Timestamp::MinusInfinity() + : Timestamp::Micros(packet_time_us)); if (!parsed_packet.Parse(std::move(packet))) { RTC_LOG(LS_ERROR) << "Failed to parse the incoming RTP packet before demuxing. Drop it."; diff --git a/pc/rtp_transport_internal.h b/pc/rtp_transport_internal.h index 4114fa9340..483a1cee38 100644 --- a/pc/rtp_transport_internal.h +++ b/pc/rtp_transport_internal.h @@ -72,7 +72,7 @@ class RtpTransportInternal : public sigslot::has_slots<> { // Called whenever a RTP packet that can not be demuxed by the transport is // received. void SetUnDemuxableRtpPacketReceivedHandler( - absl::AnyInvocable callback) { + absl::AnyInvocable callback) { callback_undemuxable_rtp_packet_received_ = std::move(callback); } @@ -160,7 +160,7 @@ class RtpTransportInternal : public sigslot::has_slots<> { CallbackList callback_list_ready_to_send_; CallbackList callback_list_rtcp_packet_received_; - absl::AnyInvocable + absl::AnyInvocable callback_undemuxable_rtp_packet_received_ = [](RtpPacketReceived& packet) {}; CallbackList> diff --git a/pc/sctp_transport.h b/pc/sctp_transport.h index 35e7656100..076dee5318 100644 --- a/pc/sctp_transport.h +++ b/pc/sctp_transport.h @@ -61,7 +61,7 @@ class SctpTransport : public SctpTransportInterface, void Start(int local_port, int remote_port, int max_message_size); // TODO(https://bugs.webrtc.org/10629): Move functions that need - // internal() to be functions on the webrtc::SctpTransport interface, + // internal() to be functions on the SctpTransport interface, // and make the internal() function private. cricket::SctpTransportInternal* internal() { RTC_DCHECK_RUN_ON(owner_thread_); diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index 0261195bb0..04d1aff789 100644 --- a/pc/sdp_offer_answer.cc +++ b/pc/sdp_offer_answer.cc @@ -86,8 +86,7 @@ namespace webrtc { namespace { -typedef webrtc::PeerConnectionInterface::RTCOfferAnswerOptions - RTCOfferAnswerOptions; +typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions; // Error messages const char kInvalidSdp[] = "Invalid session description."; @@ -834,8 +833,8 @@ std::string GenerateRtcpCname() { } // Check if we can send `new_stream` on a PeerConnection. -bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams, - webrtc::MediaStreamInterface* new_stream) { +bool CanAddLocalMediaStream(StreamCollectionInterface* current_streams, + MediaStreamInterface* new_stream) { if (!new_stream || !current_streams) { return false; } @@ -847,7 +846,7 @@ bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams, return true; } -rtc::scoped_refptr LookupDtlsTransportByMid( +rtc::scoped_refptr LookupDtlsTransportByMid( rtc::Thread* network_thread, JsepTransportController* controller, const std::string& mid) { diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h index 8aa7040b16..88ddfe0332 100644 --- a/pc/sdp_offer_answer.h +++ b/pc/sdp_offer_answer.h @@ -674,8 +674,8 @@ class SdpOfferAnswerHandler : public SdpStateProvider { // or else the CreateBuiltinVideoBitrateAllocatorFactory() will be called. // Note that one can still choose to override this in a MediaEngine // if one wants too. - std::unique_ptr - video_bitrate_allocator_factory_ RTC_GUARDED_BY(signaling_thread()); + std::unique_ptr video_bitrate_allocator_factory_ + RTC_GUARDED_BY(signaling_thread()); // Whether we are the initial offerer on the association. This // determines the SSL role. diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc index 94ceff10ac..9a44360e8f 100644 --- a/pc/sdp_offer_answer_unittest.cc +++ b/pc/sdp_offer_answer_unittest.cc @@ -88,7 +88,7 @@ class SdpOfferAnswerTest : public ::testing::Test { Dav1dDecoderTemplateAdapter>>(), nullptr /* audio_mixer */, nullptr /* audio_processing */)) { - webrtc::metrics::Reset(); + metrics::Reset(); } std::unique_ptr CreatePeerConnection() { @@ -168,8 +168,8 @@ TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsAudioVideo) { // There is no error yet but the metrics counter will increase. EXPECT_TRUE(error.ok()); EXPECT_METRIC_EQ( - 1, webrtc::metrics::NumEvents( - "WebRTC.PeerConnection.ValidBundledPayloadTypes", false)); + 1, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes", + false)); // Tolerate codec collisions in rejected m-lines. pc = CreatePeerConnection(); @@ -178,9 +178,9 @@ TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsAudioVideo) { absl::StrReplaceAll(sdp, {{"m=video 9 ", "m=video 0 "}})); pc->SetRemoteDescription(std::move(rejected_offer), &error); EXPECT_TRUE(error.ok()); - EXPECT_METRIC_EQ(1, - webrtc::metrics::NumEvents( - "WebRTC.PeerConnection.ValidBundledPayloadTypes", true)); + EXPECT_METRIC_EQ( + 1, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes", + true)); } TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsVideoFmtp) { @@ -221,8 +221,8 @@ TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsVideoFmtp) { pc->SetRemoteDescription(std::move(desc), &error); EXPECT_TRUE(error.ok()); EXPECT_METRIC_EQ( - 1, webrtc::metrics::NumEvents( - "WebRTC.PeerConnection.ValidBundledPayloadTypes", false)); + 1, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes", + false)); } TEST_F(SdpOfferAnswerTest, BundleCodecCollisionInDifferentBundlesAllowed) { @@ -264,8 +264,8 @@ TEST_F(SdpOfferAnswerTest, BundleCodecCollisionInDifferentBundlesAllowed) { pc->SetRemoteDescription(std::move(desc), &error); EXPECT_TRUE(error.ok()); EXPECT_METRIC_EQ( - 0, webrtc::metrics::NumEvents( - "WebRTC.PeerConnection.ValidBundledPayloadTypes", false)); + 0, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes", + false)); } TEST_F(SdpOfferAnswerTest, BundleMeasuresHeaderExtensionIdCollision) { diff --git a/pc/slow_peer_connection_integration_test.cc b/pc/slow_peer_connection_integration_test.cc index fd9d3417df..4e26283395 100644 --- a/pc/slow_peer_connection_integration_test.cc +++ b/pc/slow_peer_connection_integration_test.cc @@ -67,7 +67,7 @@ class FakeClockForTest : public rtc::ScopedFakeClock { // Some things use a time of "0" as a special value, so we need to start out // the fake clock at a nonzero time. // TODO(deadbeef): Fix this. - AdvanceTime(webrtc::TimeDelta::Seconds(1000)); + AdvanceTime(TimeDelta::Seconds(1000)); } // Explicit handle. @@ -170,20 +170,20 @@ TEST_P(PeerConnectionIntegrationTest, CreateTurnServer(turn_server_internal_address, turn_server_external_address, cricket::PROTO_TLS, "88.88.88.0"); - webrtc::PeerConnectionInterface::IceServer ice_server; + PeerConnectionInterface::IceServer ice_server; ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp"); ice_server.username = "test"; ice_server.password = "test"; PeerConnectionInterface::RTCConfiguration client_1_config; client_1_config.servers.push_back(ice_server); - client_1_config.type = webrtc::PeerConnectionInterface::kRelay; + client_1_config.type = PeerConnectionInterface::kRelay; PeerConnectionInterface::RTCConfiguration client_2_config; client_2_config.servers.push_back(ice_server); // Setting the type to kRelay forces the connection to go through a TURN // server. - client_2_config.type = webrtc::PeerConnectionInterface::kRelay; + client_2_config.type = PeerConnectionInterface::kRelay; // Get a copy to the pointer so we can verify calls later. rtc::TestCertificateVerifier* client_1_cert_verifier = @@ -194,10 +194,10 @@ TEST_P(PeerConnectionIntegrationTest, client_2_cert_verifier->verify_certificate_ = false; // Create the dependencies with the test certificate verifier. - webrtc::PeerConnectionDependencies client_1_deps(nullptr); + PeerConnectionDependencies client_1_deps(nullptr); client_1_deps.tls_cert_verifier = std::unique_ptr(client_1_cert_verifier); - webrtc::PeerConnectionDependencies client_2_deps(nullptr); + PeerConnectionDependencies client_2_deps(nullptr); client_2_deps.tls_cert_verifier = std::unique_ptr(client_2_cert_verifier); diff --git a/pc/srtp_transport_unittest.cc b/pc/srtp_transport_unittest.cc index fead5d6a8b..de4ff03179 100644 --- a/pc/srtp_transport_unittest.cc +++ b/pc/srtp_transport_unittest.cc @@ -342,7 +342,7 @@ class SrtpTransportTest : public ::testing::Test, public sigslot::has_slots<> { TransportObserver rtp_sink2_; int sequence_number_ = 0; - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; }; class SrtpTransportTestWithExternalAuth diff --git a/pc/test/android_test_initializer.cc b/pc/test/android_test_initializer.cc index 963544cb4b..88b4587789 100644 --- a/pc/test/android_test_initializer.cc +++ b/pc/test/android_test_initializer.cc @@ -39,7 +39,7 @@ void EnsureInitializedOnce() { RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()"; - webrtc::JVM::Initialize(jvm); + JVM::Initialize(jvm); } } // anonymous namespace diff --git a/pc/test/fake_peer_connection_base.h b/pc/test/fake_peer_connection_base.h index a1c8dca12e..1615088e99 100644 --- a/pc/test/fake_peer_connection_base.h +++ b/pc/test/fake_peer_connection_base.h @@ -363,7 +363,7 @@ class FakePeerConnectionBase : public PeerConnectionInternal { const FieldTrialsView& trials() const override { return field_trials_; } protected: - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; }; } // namespace webrtc diff --git a/pc/test/fake_peer_connection_for_stats.h b/pc/test/fake_peer_connection_for_stats.h index a65e6d5f47..33a5361394 100644 --- a/pc/test/fake_peer_connection_for_stats.h +++ b/pc/test/fake_peer_connection_for_stats.h @@ -150,7 +150,7 @@ class VoiceChannelForTesting : public cricket::VoiceChannel { receive_channel, const std::string& content_name, bool srtp_required, - webrtc::CryptoOptions crypto_options, + CryptoOptions crypto_options, rtc::UniqueRandomIdGenerator* ssrc_generator, std::string transport_name) : VoiceChannel(worker_thread, @@ -183,7 +183,7 @@ class VideoChannelForTesting : public cricket::VideoChannel { receive_channel, const std::string& content_name, bool srtp_required, - webrtc::CryptoOptions crypto_options, + CryptoOptions crypto_options, rtc::UniqueRandomIdGenerator* ssrc_generator, std::string transport_name) : VideoChannel(worker_thread, @@ -298,7 +298,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase { worker_thread_, network_thread_, signaling_thread_, std::move(voice_media_send_channel), std::move(voice_media_receive_channel), mid, kDefaultSrtpRequired, - webrtc::CryptoOptions(), context_->ssrc_generator(), transport_name); + CryptoOptions(), context_->ssrc_generator(), transport_name); auto transceiver = GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO) ->internal(); @@ -332,7 +332,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase { worker_thread_, network_thread_, signaling_thread_, std::move(video_media_send_channel), std::move(video_media_receive_channel), mid, kDefaultSrtpRequired, - webrtc::CryptoOptions(), context_->ssrc_generator(), transport_name); + CryptoOptions(), context_->ssrc_generator(), transport_name); auto transceiver = GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO) ->internal(); diff --git a/pc/test/fake_periodic_video_source.h b/pc/test/fake_periodic_video_source.h index 452a8f6c30..65652bdf0d 100644 --- a/pc/test/fake_periodic_video_source.h +++ b/pc/test/fake_periodic_video_source.h @@ -65,12 +65,12 @@ class FakePeriodicVideoSource final return wants_; } - void RemoveSink(rtc::VideoSinkInterface* sink) override { + void RemoveSink(rtc::VideoSinkInterface* sink) override { RTC_DCHECK(thread_checker_.IsCurrent()); broadcaster_.RemoveSink(sink); } - void AddOrUpdateSink(rtc::VideoSinkInterface* sink, + void AddOrUpdateSink(rtc::VideoSinkInterface* sink, const rtc::VideoSinkWants& wants) override { RTC_DCHECK(thread_checker_.IsCurrent()); { diff --git a/pc/test/integration_test_helpers.cc b/pc/test/integration_test_helpers.cc index ede159d744..64d8debc09 100644 --- a/pc/test/integration_test_helpers.cc +++ b/pc/test/integration_test_helpers.cc @@ -46,7 +46,7 @@ void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) { int FindFirstMediaStatsIndexByKind( const std::string& kind, - const std::vector& inbound_rtps) { + const std::vector& inbound_rtps) { for (size_t i = 0; i < inbound_rtps.size(); i++) { if (*inbound_rtps[i]->kind == kind) { return i; diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h index c61712cd5a..1ba332643d 100644 --- a/pc/test/integration_test_helpers.h +++ b/pc/test/integration_test_helpers.h @@ -177,14 +177,14 @@ void ReplaceFirstSsrc(StreamParams& stream, uint32_t ssrc); int FindFirstMediaStatsIndexByKind( const std::string& kind, - const std::vector& inbound_rtps); + const std::vector& inbound_rtps); -class TaskQueueMetronome : public webrtc::Metronome { +class TaskQueueMetronome : public Metronome { public: explicit TaskQueueMetronome(TimeDelta tick_period); ~TaskQueueMetronome() override; - // webrtc::Metronome implementation. + // Metronome implementation. void RequestCallOnNextTick(absl::AnyInvocable callback) override; TimeDelta TickPeriod() const override; @@ -207,7 +207,7 @@ class SignalingMessageReceiver { virtual ~SignalingMessageReceiver() {} }; -class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface { +class MockRtpReceiverObserver : public RtpReceiverObserverInterface { public: explicit MockRtpReceiverObserver(cricket::MediaType media_type) : expected_media_type_(media_type) {} @@ -234,14 +234,14 @@ class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface { // advertise support of any codecs. // TODO(steveanton): See how this could become a subclass of // PeerConnectionWrapper defined in peerconnectionwrapper.h. -class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, +class PeerConnectionIntegrationWrapper : public PeerConnectionObserver, public SignalingMessageReceiver { public: - webrtc::PeerConnectionFactoryInterface* pc_factory() const { + PeerConnectionFactoryInterface* pc_factory() const { return peer_connection_factory_.get(); } - webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); } + PeerConnectionInterface* pc() const { return peer_connection_.get(); } // If a signaling message receiver is set (via ConnectFakeSignaling), this // will set the whole offer/answer exchange in motion. Just need to wait for @@ -339,11 +339,11 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, return AddTrack(CreateLocalVideoTrack()); } - rtc::scoped_refptr CreateLocalAudioTrack() { + rtc::scoped_refptr CreateLocalAudioTrack() { cricket::AudioOptions options; // Disable highpass filter so that we can get all the test audio frames. options.highpass_filter = false; - rtc::scoped_refptr source = + rtc::scoped_refptr source = peer_connection_factory_->CreateAudioSource(options); // TODO(perkj): Test audio source when it is implemented. Currently audio // always use the default input. @@ -351,21 +351,20 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, source.get()); } - rtc::scoped_refptr CreateLocalVideoTrack() { - webrtc::FakePeriodicVideoSource::Config config; + rtc::scoped_refptr CreateLocalVideoTrack() { + FakePeriodicVideoSource::Config config; config.timestamp_offset_ms = rtc::TimeMillis(); return CreateLocalVideoTrackInternal(config); } - rtc::scoped_refptr - CreateLocalVideoTrackWithConfig( - webrtc::FakePeriodicVideoSource::Config config) { + rtc::scoped_refptr CreateLocalVideoTrackWithConfig( + FakePeriodicVideoSource::Config config) { return CreateLocalVideoTrackInternal(config); } - rtc::scoped_refptr - CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) { - webrtc::FakePeriodicVideoSource::Config config; + rtc::scoped_refptr CreateLocalVideoTrackWithRotation( + VideoRotation rotation) { + FakePeriodicVideoSource::Config config; config.rotation = rotation; config.timestamp_offset_ms = rtc::TimeMillis(); return CreateLocalVideoTrackInternal(config); @@ -409,22 +408,22 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, } bool SignalingStateStable() { - return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable; + return pc()->signaling_state() == PeerConnectionInterface::kStable; } bool IceGatheringStateComplete() { return pc()->ice_gathering_state() == - webrtc::PeerConnectionInterface::kIceGatheringComplete; + PeerConnectionInterface::kIceGatheringComplete; } void CreateDataChannel() { CreateDataChannel(nullptr); } - void CreateDataChannel(const webrtc::DataChannelInit* init) { + void CreateDataChannel(const DataChannelInit* init) { CreateDataChannel(kDataChannelLabel, init); } void CreateDataChannel(const std::string& label, - const webrtc::DataChannelInit* init) { + const DataChannelInit* init) { auto data_channel_or_error = pc()->CreateDataChannelOrError(label, init); ASSERT_TRUE(data_channel_or_error.ok()); data_channels_.push_back(data_channel_or_error.MoveValue()); @@ -482,7 +481,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, // Returns a MockStatsObserver in a state after stats gathering finished, // which can be used to access the gathered stats. rtc::scoped_refptr OldGetStatsForTrack( - webrtc::MediaStreamTrackInterface* track) { + MediaStreamTrackInterface* track) { auto observer = rtc::make_ref_counted(); EXPECT_TRUE(peer_connection_->GetStats( observer.get(), nullptr, @@ -498,9 +497,8 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, // Synchronously gets stats and returns them. If it times out, fails the test // and returns null. - rtc::scoped_refptr NewGetStats() { - auto callback = - rtc::make_ref_counted(); + rtc::scoped_refptr NewGetStats() { + auto callback = rtc::make_ref_counted(); peer_connection_->GetStats(callback.get()); EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout); return callback->report(); @@ -527,10 +525,10 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, return static_cast(rendered_width()) / rendered_height(); } - webrtc::VideoRotation rendered_rotation() { + VideoRotation rendered_rotation() { EXPECT_FALSE(fake_video_renderers_.empty()); return fake_video_renderers_.empty() - ? webrtc::kVideoRotation_0 + ? kVideoRotation_0 : fake_video_renderers_.begin()->second->rotation(); } @@ -573,20 +571,20 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, return pc()->local_streams().get(); } - webrtc::PeerConnectionInterface::SignalingState signaling_state() { + PeerConnectionInterface::SignalingState signaling_state() { return pc()->signaling_state(); } - webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() { + PeerConnectionInterface::IceConnectionState ice_connection_state() { return pc()->ice_connection_state(); } - webrtc::PeerConnectionInterface::IceConnectionState + PeerConnectionInterface::IceConnectionState standardized_ice_connection_state() { return pc()->standardized_ice_connection_state(); } - webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() { + PeerConnectionInterface::IceGatheringState ice_gathering_state() { return pc()->ice_gathering_state(); } @@ -615,7 +613,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, } cricket::PortAllocator* port_allocator() const { return port_allocator_; } - webrtc::FakeRtcEventLogFactory* event_log_factory() const { + FakeRtcEventLogFactory* event_log_factory() const { return event_log_factory_; } @@ -628,8 +626,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, // Sets the mDNS responder for the owned fake network manager and keeps a // reference to the responder. - void SetMdnsResponder( - std::unique_ptr mdns_responder) { + void SetMdnsResponder(std::unique_ptr mdns_responder) { RTC_DCHECK(mdns_responder != nullptr); mdns_responder_ = mdns_responder.get(); network_manager()->set_mdns_responder(std::move(mdns_responder)); @@ -644,7 +641,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, } bool Rollback() { return SetRemoteDescription( - webrtc::CreateSessionDescription(SdpType::kRollback, "")); + CreateSessionDescription(SdpType::kRollback, "")); } // Functions for querying stats. @@ -652,7 +649,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, // Get the baseline numbers for audio_packets and audio_delay. auto received_stats = NewGetStats(); auto rtp_stats = - received_stats->GetStatsOfType()[0]; + received_stats->GetStatsOfType()[0]; ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.is_defined()); ASSERT_TRUE(rtp_stats->packets_received.is_defined()); rtp_stats_id_ = rtp_stats->id(); @@ -664,8 +661,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, void UpdateDelayStats(std::string tag, int desc_size) { auto report = NewGetStats(); - auto rtp_stats = - report->GetAs(rtp_stats_id_); + auto rtp_stats = report->GetAs(rtp_stats_id_); ASSERT_TRUE(rtp_stats); auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_; auto delta_rpad = @@ -744,11 +740,11 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, bool Init(const PeerConnectionFactory::Options* options, const PeerConnectionInterface::RTCConfiguration* config, - webrtc::PeerConnectionDependencies dependencies, + PeerConnectionDependencies dependencies, rtc::SocketServer* socket_server, rtc::Thread* network_thread, rtc::Thread* worker_thread, - std::unique_ptr event_log_factory, + std::unique_ptr event_log_factory, bool reset_encoder_factory, bool reset_decoder_factory, bool create_media_engine) { @@ -771,12 +767,12 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, } rtc::Thread* const signaling_thread = rtc::Thread::Current(); - webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies; + PeerConnectionFactoryDependencies pc_factory_dependencies; pc_factory_dependencies.network_thread = network_thread; pc_factory_dependencies.worker_thread = worker_thread; pc_factory_dependencies.signaling_thread = signaling_thread; pc_factory_dependencies.task_queue_factory = - webrtc::CreateDefaultTaskQueueFactory(); + CreateDefaultTaskQueueFactory(); pc_factory_dependencies.trials = std::make_unique(); pc_factory_dependencies.metronome = std::make_unique(TimeDelta::Millis(8)); @@ -805,11 +801,11 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, pc_factory_dependencies.event_log_factory = std::move(event_log_factory); } else { pc_factory_dependencies.event_log_factory = - std::make_unique( + std::make_unique( pc_factory_dependencies.task_queue_factory.get()); } - peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory( - std::move(pc_factory_dependencies)); + peer_connection_factory_ = + CreateModularPeerConnectionFactory(std::move(pc_factory_dependencies)); if (!peer_connection_factory_) { return false; @@ -826,9 +822,9 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, return peer_connection_.get() != nullptr; } - rtc::scoped_refptr CreatePeerConnection( + rtc::scoped_refptr CreatePeerConnection( const PeerConnectionInterface::RTCConfiguration* config, - webrtc::PeerConnectionDependencies dependencies) { + PeerConnectionDependencies dependencies) { PeerConnectionInterface::RTCConfiguration modified_config; modified_config.sdp_semantics = sdp_semantics_; // If `config` is null, this will result in a default configuration being @@ -861,21 +857,20 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, signal_ice_candidates_ = signal; } - rtc::scoped_refptr CreateLocalVideoTrackInternal( - webrtc::FakePeriodicVideoSource::Config config) { + rtc::scoped_refptr CreateLocalVideoTrackInternal( + FakePeriodicVideoSource::Config config) { // Set max frame rate to 10fps to reduce the risk of test flakiness. // TODO(deadbeef): Do something more robust. config.frame_interval_ms = 100; video_track_sources_.emplace_back( - rtc::make_ref_counted( + rtc::make_ref_counted( config, false /* remote */)); - rtc::scoped_refptr track = + rtc::scoped_refptr track = peer_connection_factory_->CreateVideoTrack(video_track_sources_.back(), rtc::CreateRandomUuid()); if (!local_video_renderer_) { - local_video_renderer_.reset( - new webrtc::FakeVideoTrackRenderer(track.get())); + local_video_renderer_.reset(new FakeVideoTrackRenderer(track.get())); } return track; } @@ -883,7 +878,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, void HandleIncomingOffer(const std::string& msg) { RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer"; std::unique_ptr desc = - webrtc::CreateSessionDescription(SdpType::kOffer, msg); + CreateSessionDescription(SdpType::kOffer, msg); if (received_sdp_munger_) { received_sdp_munger_(desc->description()); } @@ -903,7 +898,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, void HandleIncomingAnswer(const std::string& msg) { RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer"; std::unique_ptr desc = - webrtc::CreateSessionDescription(SdpType::kAnswer, msg); + CreateSessionDescription(SdpType::kAnswer, msg); if (received_sdp_munger_) { received_sdp_munger_(desc->description()); } @@ -1054,7 +1049,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, const std::string& msg) override { RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage"; absl::optional result; - pc()->AddIceCandidate(absl::WrapUnique(webrtc::CreateIceCandidate( + pc()->AddIceCandidate(absl::WrapUnique(CreateIceCandidate( sdp_mid, sdp_mline_index, msg, nullptr)), [&result](RTCError r) { result = r; }); EXPECT_TRUE_WAIT(result.has_value(), kDefaultTimeout); @@ -1063,7 +1058,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, // PeerConnectionObserver callbacks. void OnSignalingChange( - webrtc::PeerConnectionInterface::SignalingState new_state) override { + PeerConnectionInterface::SignalingState new_state) override { EXPECT_EQ(pc()->signaling_state(), new_state); peer_connection_signaling_state_history_.push_back(new_state); } @@ -1092,21 +1087,21 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, } void OnRenegotiationNeeded() override {} void OnIceConnectionChange( - webrtc::PeerConnectionInterface::IceConnectionState new_state) override { + PeerConnectionInterface::IceConnectionState new_state) override { EXPECT_EQ(pc()->ice_connection_state(), new_state); ice_connection_state_history_.push_back(new_state); } void OnStandardizedIceConnectionChange( - webrtc::PeerConnectionInterface::IceConnectionState new_state) override { + PeerConnectionInterface::IceConnectionState new_state) override { standardized_ice_connection_state_history_.push_back(new_state); } void OnConnectionChange( - webrtc::PeerConnectionInterface::PeerConnectionState new_state) override { + PeerConnectionInterface::PeerConnectionState new_state) override { peer_connection_state_history_.push_back(new_state); } void OnIceGatheringChange( - webrtc::PeerConnectionInterface::IceGatheringState new_state) override { + PeerConnectionInterface::IceGatheringState new_state) override { EXPECT_EQ(pc()->ice_gathering_state(), new_state); ice_gathering_state_history_.push_back(new_state); } @@ -1116,7 +1111,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, ice_candidate_pair_change_history_.push_back(event); } - void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override { + void OnIceCandidate(const IceCandidateInterface* candidate) override { RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate"; if (remote_async_dns_resolver_) { @@ -1172,20 +1167,19 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, std::unique_ptr fake_network_manager_; std::unique_ptr socket_factory_; // Reference to the mDNS responder owned by `fake_network_manager_` after set. - webrtc::FakeMdnsResponder* mdns_responder_ = nullptr; + FakeMdnsResponder* mdns_responder_ = nullptr; - rtc::scoped_refptr peer_connection_; - rtc::scoped_refptr - peer_connection_factory_; + rtc::scoped_refptr peer_connection_; + rtc::scoped_refptr peer_connection_factory_; cricket::PortAllocator* port_allocator_; // Needed to keep track of number of frames sent. rtc::scoped_refptr fake_audio_capture_module_; // Needed to keep track of number of frames received. - std::map> + std::map> fake_video_renderers_; // Needed to ensure frames aren't received for removed tracks. - std::vector> + std::vector> removed_fake_video_renderers_; // For remote peer communication. @@ -1197,10 +1191,9 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, // Store references to the video sources we've created, so that we can stop // them, if required. - std::vector> - video_track_sources_; + std::vector> video_track_sources_; // `local_video_renderer_` attached to the first created local video track. - std::unique_ptr local_video_renderer_; + std::unique_ptr local_video_renderer_; SdpSemantics sdp_semantics_; PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_; @@ -1230,7 +1223,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, ice_candidate_pair_change_history_; std::vector peer_connection_signaling_state_history_; - webrtc::FakeRtcEventLogFactory* event_log_factory_; + FakeRtcEventLogFactory* event_log_factory_; // Number of ICE candidates expected. The default is no limit. int candidates_expected_ = std::numeric_limits::max(); @@ -1247,7 +1240,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, friend class PeerConnectionIntegrationBaseTest; }; -class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput { +class MockRtcEventLogOutput : public RtcEventLogOutput { public: virtual ~MockRtcEventLogOutput() = default; MOCK_METHOD(bool, IsActive, (), (const, override)); @@ -1359,7 +1352,7 @@ class MediaExpectations { int callee_video_frames_expected_ = 0; }; -class MockIceTransport : public webrtc::IceTransportInterface { +class MockIceTransport : public IceTransportInterface { public: MockIceTransport(const std::string& name, int component) : internal_(std::make_unique( @@ -1407,7 +1400,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { worker_thread_->SetName("PCWorkerThread", this); RTC_CHECK(network_thread_->Start()); RTC_CHECK(worker_thread_->Start()); - webrtc::metrics::Reset(); + metrics::Reset(); } ~PeerConnectionIntegrationBaseTest() { @@ -1444,13 +1437,13 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { // are connected. This is an important distinction. Once we have separate // ICE and DTLS state, this check needs to use the DTLS state. return (callee()->ice_connection_state() == - webrtc::PeerConnectionInterface::kIceConnectionConnected || + PeerConnectionInterface::kIceConnectionConnected || callee()->ice_connection_state() == - webrtc::PeerConnectionInterface::kIceConnectionCompleted) && + PeerConnectionInterface::kIceConnectionCompleted) && (caller()->ice_connection_state() == - webrtc::PeerConnectionInterface::kIceConnectionConnected || + PeerConnectionInterface::kIceConnectionConnected || caller()->ice_connection_state() == - webrtc::PeerConnectionInterface::kIceConnectionCompleted); + PeerConnectionInterface::kIceConnectionCompleted); } // When `event_log_factory` is null, the default implementation of the event @@ -1459,8 +1452,8 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { const std::string& debug_name, const PeerConnectionFactory::Options* options, const RTCConfiguration* config, - webrtc::PeerConnectionDependencies dependencies, - std::unique_ptr event_log_factory, + PeerConnectionDependencies dependencies, + std::unique_ptr event_log_factory, bool reset_encoder_factory, bool reset_decoder_factory, bool create_media_engine = true) { @@ -1490,10 +1483,10 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { const std::string& debug_name, const PeerConnectionFactory::Options* options, const RTCConfiguration* config, - webrtc::PeerConnectionDependencies dependencies) { + PeerConnectionDependencies dependencies) { return CreatePeerConnectionWrapper( debug_name, options, config, std::move(dependencies), - std::make_unique(), + std::make_unique(), /*reset_encoder_factory=*/false, /*reset_decoder_factory=*/false); } @@ -1514,17 +1507,17 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { // callee PeerConnections. SdpSemantics original_semantics = sdp_semantics_; sdp_semantics_ = caller_semantics; - caller_ = CreatePeerConnectionWrapper( - "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), - nullptr, - /*reset_encoder_factory=*/false, - /*reset_decoder_factory=*/false); + caller_ = CreatePeerConnectionWrapper("Caller", nullptr, nullptr, + PeerConnectionDependencies(nullptr), + nullptr, + /*reset_encoder_factory=*/false, + /*reset_decoder_factory=*/false); sdp_semantics_ = callee_semantics; - callee_ = CreatePeerConnectionWrapper( - "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), - nullptr, - /*reset_encoder_factory=*/false, - /*reset_decoder_factory=*/false); + callee_ = CreatePeerConnectionWrapper("Callee", nullptr, nullptr, + PeerConnectionDependencies(nullptr), + nullptr, + /*reset_encoder_factory=*/false, + /*reset_decoder_factory=*/false); sdp_semantics_ = original_semantics; return caller_ && callee_; } @@ -1532,24 +1525,24 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { bool CreatePeerConnectionWrappersWithConfig( const PeerConnectionInterface::RTCConfiguration& caller_config, const PeerConnectionInterface::RTCConfiguration& callee_config) { - caller_ = CreatePeerConnectionWrapper( - "Caller", nullptr, &caller_config, - webrtc::PeerConnectionDependencies(nullptr), nullptr, - /*reset_encoder_factory=*/false, - /*reset_decoder_factory=*/false); - callee_ = CreatePeerConnectionWrapper( - "Callee", nullptr, &callee_config, - webrtc::PeerConnectionDependencies(nullptr), nullptr, - /*reset_encoder_factory=*/false, - /*reset_decoder_factory=*/false); + caller_ = CreatePeerConnectionWrapper("Caller", nullptr, &caller_config, + PeerConnectionDependencies(nullptr), + nullptr, + /*reset_encoder_factory=*/false, + /*reset_decoder_factory=*/false); + callee_ = CreatePeerConnectionWrapper("Callee", nullptr, &callee_config, + PeerConnectionDependencies(nullptr), + nullptr, + /*reset_encoder_factory=*/false, + /*reset_decoder_factory=*/false); return caller_ && callee_; } bool CreatePeerConnectionWrappersWithConfigAndDeps( const PeerConnectionInterface::RTCConfiguration& caller_config, - webrtc::PeerConnectionDependencies caller_dependencies, + PeerConnectionDependencies caller_dependencies, const PeerConnectionInterface::RTCConfiguration& callee_config, - webrtc::PeerConnectionDependencies callee_dependencies) { + PeerConnectionDependencies callee_dependencies) { caller_ = CreatePeerConnectionWrapper("Caller", nullptr, &caller_config, std::move(caller_dependencies), nullptr, @@ -1566,16 +1559,16 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { bool CreatePeerConnectionWrappersWithOptions( const PeerConnectionFactory::Options& caller_options, const PeerConnectionFactory::Options& callee_options) { - caller_ = CreatePeerConnectionWrapper( - "Caller", &caller_options, nullptr, - webrtc::PeerConnectionDependencies(nullptr), nullptr, - /*reset_encoder_factory=*/false, - /*reset_decoder_factory=*/false); - callee_ = CreatePeerConnectionWrapper( - "Callee", &callee_options, nullptr, - webrtc::PeerConnectionDependencies(nullptr), nullptr, - /*reset_encoder_factory=*/false, - /*reset_decoder_factory=*/false); + caller_ = CreatePeerConnectionWrapper("Caller", &caller_options, nullptr, + PeerConnectionDependencies(nullptr), + nullptr, + /*reset_encoder_factory=*/false, + /*reset_decoder_factory=*/false); + callee_ = CreatePeerConnectionWrapper("Callee", &callee_options, nullptr, + PeerConnectionDependencies(nullptr), + nullptr, + /*reset_encoder_factory=*/false, + /*reset_decoder_factory=*/false); return caller_ && callee_; } @@ -1583,10 +1576,10 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { PeerConnectionInterface::RTCConfiguration default_config; caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog( "Caller", nullptr, &default_config, - webrtc::PeerConnectionDependencies(nullptr)); + PeerConnectionDependencies(nullptr)); callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog( "Callee", nullptr, &default_config, - webrtc::PeerConnectionDependencies(nullptr)); + PeerConnectionDependencies(nullptr)); return caller_ && callee_; } @@ -1596,7 +1589,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { new FakeRTCCertificateGenerator()); cert_generator->use_alternate_key(); - webrtc::PeerConnectionDependencies dependencies(nullptr); + PeerConnectionDependencies dependencies(nullptr); dependencies.cert_generator = std::move(cert_generator); return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr, std::move(dependencies), nullptr, @@ -1606,12 +1599,12 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { bool CreateOneDirectionalPeerConnectionWrappers(bool caller_to_callee) { caller_ = CreatePeerConnectionWrapper( - "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), + "Caller", nullptr, nullptr, PeerConnectionDependencies(nullptr), nullptr, /*reset_encoder_factory=*/!caller_to_callee, /*reset_decoder_factory=*/caller_to_callee); callee_ = CreatePeerConnectionWrapper( - "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), + "Callee", nullptr, nullptr, PeerConnectionDependencies(nullptr), nullptr, /*reset_encoder_factory=*/caller_to_callee, /*reset_decoder_factory=*/!caller_to_callee); @@ -1619,18 +1612,18 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { } bool CreatePeerConnectionWrappersWithoutMediaEngine() { - caller_ = CreatePeerConnectionWrapper( - "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), - nullptr, - /*reset_encoder_factory=*/false, - /*reset_decoder_factory=*/false, - /*create_media_engine=*/false); - callee_ = CreatePeerConnectionWrapper( - "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), - nullptr, - /*reset_encoder_factory=*/false, - /*reset_decoder_factory=*/false, - /*create_media_engine=*/false); + caller_ = CreatePeerConnectionWrapper("Caller", nullptr, nullptr, + PeerConnectionDependencies(nullptr), + nullptr, + /*reset_encoder_factory=*/false, + /*reset_decoder_factory=*/false, + /*create_media_engine=*/false); + callee_ = CreatePeerConnectionWrapper("Callee", nullptr, nullptr, + PeerConnectionDependencies(nullptr), + nullptr, + /*reset_encoder_factory=*/false, + /*reset_decoder_factory=*/false, + /*create_media_engine=*/false); return caller_ && callee_; } @@ -1700,7 +1693,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { // Messages may get lost on the unreliable DataChannel, so we send multiple // times to avoid test flakiness. - void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc, + void SendRtpDataWithRetries(DataChannelInterface* dc, const std::string& data, int retries) { for (int i = 0; i < retries; ++i) { diff --git a/pc/test/mock_peer_connection_observers.h b/pc/test/mock_peer_connection_observers.h index e9d97a97f6..6222ef7719 100644 --- a/pc/test/mock_peer_connection_observers.h +++ b/pc/test/mock_peer_connection_observers.h @@ -254,7 +254,7 @@ class MockPeerConnectionObserver : public PeerConnectionObserver { }; class MockCreateSessionDescriptionObserver - : public webrtc::CreateSessionDescriptionObserver { + : public CreateSessionDescriptionObserver { public: MockCreateSessionDescriptionObserver() : called_(false), @@ -266,7 +266,7 @@ class MockCreateSessionDescriptionObserver error_ = ""; desc_.reset(desc); } - void OnFailure(webrtc::RTCError error) override { + void OnFailure(RTCError error) override { MutexLock lock(&mutex_); called_ = true; error_ = error.message(); @@ -295,8 +295,7 @@ class MockCreateSessionDescriptionObserver std::unique_ptr desc_ RTC_GUARDED_BY(mutex_); }; -class MockSetSessionDescriptionObserver - : public webrtc::SetSessionDescriptionObserver { +class MockSetSessionDescriptionObserver : public SetSessionDescriptionObserver { public: static rtc::scoped_refptr Create() { return rtc::make_ref_counted(); @@ -312,7 +311,7 @@ class MockSetSessionDescriptionObserver called_ = true; error_ = ""; } - void OnFailure(webrtc::RTCError error) override { + void OnFailure(RTCError error) override { MutexLock lock(&mutex_); called_ = true; error_ = error.message(); @@ -375,14 +374,14 @@ class FakeSetRemoteDescriptionObserver absl::optional error_; }; -class MockDataChannelObserver : public webrtc::DataChannelObserver { +class MockDataChannelObserver : public DataChannelObserver { public: struct Message { std::string data; bool binary; }; - explicit MockDataChannelObserver(webrtc::DataChannelInterface* channel) + explicit MockDataChannelObserver(DataChannelInterface* channel) : channel_(channel) { channel_->RegisterObserver(this); states_.push_back(channel_->state()); @@ -419,12 +418,12 @@ class MockDataChannelObserver : public webrtc::DataChannelObserver { } private: - rtc::scoped_refptr channel_; + rtc::scoped_refptr channel_; std::vector states_; std::vector messages_; }; -class MockStatsObserver : public webrtc::StatsObserver { +class MockStatsObserver : public StatsObserver { public: MockStatsObserver() : called_(false), stats_() {} virtual ~MockStatsObserver() {} @@ -576,7 +575,7 @@ class MockStatsObserver : public webrtc::StatsObserver { }; // Helper class that just stores the report from the callback. -class MockRTCStatsCollectorCallback : public webrtc::RTCStatsCollectorCallback { +class MockRTCStatsCollectorCallback : public RTCStatsCollectorCallback { public: rtc::scoped_refptr report() { return report_; } diff --git a/pc/test/rtp_transport_test_util.h b/pc/test/rtp_transport_test_util.h index 8aeaf07c1d..563014f94a 100644 --- a/pc/test/rtp_transport_test_util.h +++ b/pc/test/rtp_transport_test_util.h @@ -33,9 +33,7 @@ class TransportObserver : public RtpPacketSinkInterface { rtp_transport->SubscribeReadyToSend( this, [this](bool arg) { OnReadyToSend(arg); }); rtp_transport->SetUnDemuxableRtpPacketReceivedHandler( - [this](webrtc::RtpPacketReceived& packet) { - OnUndemuxableRtpPacket(packet); - }); + [this](RtpPacketReceived& packet) { OnUndemuxableRtpPacket(packet); }); rtp_transport->SubscribeSentPacket(this, [this](const rtc::SentPacket& packet) { sent_packet_count_++; diff --git a/pc/test/svc_e2e_tests.cc b/pc/test/svc_e2e_tests.cc index ae35c7f676..412027bba1 100644 --- a/pc/test/svc_e2e_tests.cc +++ b/pc/test/svc_e2e_tests.cc @@ -160,10 +160,9 @@ std::string SvcTestNameGenerator( // encoder and decoder level. class SvcVideoQualityAnalyzer : public DefaultVideoQualityAnalyzer { public: - using SpatialTemporalLayerCounts = - webrtc::flat_map>; + using SpatialTemporalLayerCounts = flat_map>; - explicit SvcVideoQualityAnalyzer(webrtc::Clock* clock) + explicit SvcVideoQualityAnalyzer(Clock* clock) : DefaultVideoQualityAnalyzer(clock, test::GetGlobalMetricsLogger(), DefaultVideoQualityAnalyzerOptions{ @@ -315,9 +314,9 @@ TEST_P(SvcTest, ScalabilityModeSupported) { if (UseDependencyDescriptor()) { trials += "WebRTC-DependencyDescriptorAdvertised/Enabled/"; } - webrtc::test::ScopedFieldTrials override_trials(AppendFieldTrials(trials)); + test::ScopedFieldTrials override_trials(AppendFieldTrials(trials)); std::unique_ptr network_emulation_manager = - CreateNetworkEmulationManager(webrtc::TimeMode::kSimulated); + CreateNetworkEmulationManager(TimeMode::kSimulated); auto analyzer = std::make_unique( network_emulation_manager->time_controller()->GetClock()); SvcVideoQualityAnalyzer* analyzer_ptr = analyzer.get(); diff --git a/pc/video_rtp_receiver_unittest.cc b/pc/video_rtp_receiver_unittest.cc index 5ff736084f..e9729170b7 100644 --- a/pc/video_rtp_receiver_unittest.cc +++ b/pc/video_rtp_receiver_unittest.cc @@ -94,7 +94,7 @@ class VideoRtpReceiverTest : public testing::Test { [&]() { receiver_->SetMediaChannel(media_channel); }); } - webrtc::VideoTrackSourceInterface* Source() { + VideoTrackSourceInterface* Source() { return receiver_->streams()[0]->FindVideoTrack("receiver")->GetSource(); } diff --git a/pc/video_rtp_track_source_unittest.cc b/pc/video_rtp_track_source_unittest.cc index 13728c7eff..55632cea42 100644 --- a/pc/video_rtp_track_source_unittest.cc +++ b/pc/video_rtp_track_source_unittest.cc @@ -109,11 +109,11 @@ TEST(VideoRtpTrackSourceTest, NoCallbacksAfterClearedCallback) { class TestFrame : public RecordableEncodedFrame { public: - rtc::scoped_refptr encoded_buffer() + rtc::scoped_refptr encoded_buffer() const override { return nullptr; } - absl::optional color_space() const override { + absl::optional color_space() const override { return absl::nullopt; } VideoCodecType codec() const override { return kVideoCodecGeneric; } diff --git a/pc/video_track.h b/pc/video_track.h index 13a51c454b..e504182c82 100644 --- a/pc/video_track.h +++ b/pc/video_track.h @@ -70,7 +70,7 @@ class VideoTrack : public MediaStreamTrack, // Implements ObserverInterface. Observes `video_source_` state. void OnChanged() override; - RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker signaling_thread_; + RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_; rtc::Thread* const worker_thread_; const rtc::scoped_refptr< VideoTrackSourceProxyWithInternal> diff --git a/pc/video_track_source_proxy.h b/pc/video_track_source_proxy.h index 8500a98766..40d24234ad 100644 --- a/pc/video_track_source_proxy.h +++ b/pc/video_track_source_proxy.h @@ -52,7 +52,7 @@ PROXY_SECONDARY_METHOD1(void, rtc::VideoSinkInterface*) PROXY_SECONDARY_METHOD1(void, ProcessConstraints, - const webrtc::VideoTrackSourceConstraints&) + const VideoTrackSourceConstraints&) END_PROXY_MAP(VideoTrackSource) } // namespace webrtc diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc index 7d896e25bc..da024eab81 100644 --- a/pc/webrtc_sdp.cc +++ b/pc/webrtc_sdp.cc @@ -2626,7 +2626,7 @@ static std::unique_ptr ParseContentDescription( int* msid_signaling, TransportDescription* transport, std::vector>* candidates, - webrtc::SdpParseError* error) { + SdpParseError* error) { std::unique_ptr media_desc; if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) { media_desc = std::make_unique(); diff --git a/tools_webrtc/remove_extra_namespace.py b/tools_webrtc/remove_extra_namespace.py new file mode 100755 index 0000000000..21ac2d1aa2 --- /dev/null +++ b/tools_webrtc/remove_extra_namespace.py @@ -0,0 +1,93 @@ +#!/usr/bin/env vpython3 + +# Copyright (c) 2023 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. +"""Remove extra namespace qualifications + +Looks for names that don't need to be qualified by namespace, and deletes +the qualifier. + +Depends on namespace names being properly formatted +""" +import os +import glob +import sys +import re +import argparse + + +def remove_extra_namespace_from_file(namespace, filename): + print('Processing namespace', namespace, 'file', filename) + with open(filename) as file: + newfile = open(filename + '.NEW', 'w') + namespaces = [] + changes = 0 + for line in file: + match = re.match(r'namespace (\S+) {', line) + if match is not None: + namespaces.insert(0, match.group(1)) + newfile.write(line) + continue + match = re.match(r'}\s+// namespace (\S+)$', line) + if match is not None: + if match.group(1) != namespaces[0]: + print('Namespace mismatch') + raise RuntimeError('Namespace mismatch') + del namespaces[0] + newfile.write(line) + continue + # Remove namespace usage. Only replacing when target + # namespace is the innermost namespace. + if len(namespaces) > 0 and namespaces[0] == namespace: + # Note that in namespace foo, we match neither ::foo::name + # nor morefoo::name + # Neither do we match foo:: when it is not followed by + # an identifier character. + usage_re = r'(?<=[^a-z:]){}::(?=[a-zA-Z])'.format( + namespaces[0]) + if re.search(usage_re, line): + line = re.sub(usage_re, '', line) + changes += 1 + newfile.write(line) + if changes > 0: + print('Made', changes, 'changes to', filename) + os.remove(filename) + os.rename(filename + '.NEW', filename) + else: + os.remove(filename + '.NEW') + + +def remove_extra_namespace_from_files(namespace, files): + for file in files: + if os.path.isfile(file): + if re.search(r'\.(h|cc)$', file): + remove_extra_namespace_from_file(namespace, file) + elif os.path.isdir(file): + if file in ('third_party', 'out'): + continue + subfiles = glob.glob(file + '/*') + remove_extra_namespace_from_files(namespace, subfiles) + else: + print(file, 'is not a file or directory, ignoring') + + +def main(): + parser = argparse.ArgumentParser( + prog='remove_extra_namespace.py', + description=__doc__.strip().splitlines()[0], + epilog=''.join(__doc__.splitlines(True)[1:]), + formatter_class=argparse.RawDescriptionHelpFormatter, + ) + parser.add_argument('--namespace') + parser.add_argument('files', nargs=argparse.REMAINDER) + args = parser.parse_args() + return remove_extra_namespace_from_files(args.namespace, args.files) + + +if __name__ == '__main__': + sys.exit(main())