/external/webrtc/webrtc/base/ |
worker.cc | 17 namespace rtc { namespace 33 rtc::Thread *me = rtc::Thread::Current(); 53 } else if (worker_thread_ != rtc::Thread::Current()) { 69 void Worker::OnMessage(rtc::Message *msg) { 71 ASSERT(worker_thread_ == rtc::Thread::Current()); 75 } // namespace rtc
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keep_ref_until_done.h | 19 namespace rtc { namespace 31 return rtc::Bind(&impl::DoNothing<ObjectT>, scoped_refptr<ObjectT>(object)); 37 return rtc::Bind(&impl::DoNothing<ObjectT>, object); 40 } // namespace rtc
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asyncpacketsocket.cc | 13 namespace rtc { namespace 29 }; // namespace rtc
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asynctcpsocket_unittest.cc | 19 namespace rtc { namespace 26 : pss_(new rtc::PhysicalSocketServer), 27 vss_(new rtc::VirtualSocketServer(pss_.get())), 35 void OnReadyToSend(rtc::AsyncPacketSocket* socket) { 53 } // namespace rtc
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asyncudpsocket_unittest.cc | 19 namespace rtc { namespace 26 : pss_(new rtc::PhysicalSocketServer), 27 vss_(new rtc::VirtualSocketServer(pss_.get())), 35 void OnReadyToSend(rtc::AsyncPacketSocket* socket) { 53 } // namespace rtc
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proxyinfo.cc | 13 namespace rtc { namespace 24 } // namespace rtc
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socketserver.h | 16 namespace rtc { namespace 44 } // namespace rtc
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sslfingerprint.h | 20 namespace rtc { namespace 26 const rtc::SSLIdentity* identity); 29 const rtc::SSLCertificate* cert); 47 rtc::Buffer digest; 50 } // namespace rtc
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thread_checker_impl.h | 19 namespace rtc { namespace 46 } // namespace rtc
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versionparsing.h | 16 namespace rtc { namespace 33 } // namespace rtc
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/external/webrtc/webrtc/modules/audio_processing/ |
level_estimator_impl.cc | 19 LevelEstimatorImpl::LevelEstimatorImpl(rtc::CriticalSection* crit) 27 rtc::CritScope cs(crit_); 33 rtc::CritScope cs(crit_); 44 rtc::CritScope cs(crit_); 53 rtc::CritScope cs(crit_); 58 rtc::CritScope cs(crit_);
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level_estimator_impl.h | 26 explicit LevelEstimatorImpl(rtc::CriticalSection* crit); 39 rtc::CriticalSection* const crit_ = nullptr; 41 rtc::scoped_ptr<RMSLevel> rms_ GUARDED_BY(crit_);
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/external/webrtc/webrtc/libjingle/xmpp/ |
xmppthread.h | 23 public rtc::Thread, buzz::XmppPumpNotify, rtc::MessageHandler { 39 void OnMessage(rtc::Message* pmsg);
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pingtask.h | 28 class PingTask : public buzz::XmppTask, private rtc::MessageHandler { 31 rtc::MessageQueue* message_queue, 44 virtual void OnMessage(rtc::Message* msg); 46 rtc::MessageQueue* message_queue_;
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/external/webrtc/webrtc/modules/audio_coding/test/ |
SpatialAudio.h | 36 rtc::scoped_ptr<AudioCodingModule> _acmLeft; 37 rtc::scoped_ptr<AudioCodingModule> _acmRight; 38 rtc::scoped_ptr<AudioCodingModule> _acmReceiver;
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TwoWayCommunication.h | 34 rtc::scoped_ptr<AudioCodingModule> _acmA; 35 rtc::scoped_ptr<AudioCodingModule> _acmB; 37 rtc::scoped_ptr<AudioCodingModule> _acmRefA; 38 rtc::scoped_ptr<AudioCodingModule> _acmRefB;
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/external/webrtc/webrtc/p2p/base/ |
portinterface.h | 20 namespace rtc { namespace 45 virtual rtc::Network* Network() const = 0; 67 const rtc::SocketAddress& remote_addr) = 0; 75 virtual int SetOption(rtc::Socket::Option opt, int value) = 0; 76 virtual int GetOption(rtc::Socket::Option opt, int* value) = 0; 84 const rtc::SocketAddress& addr, 85 const rtc::PacketOptions& options, bool payload) = 0; 90 sigslot::signal6<PortInterface*, const rtc::SocketAddress&, 98 const rtc::SocketAddress& addr) = 0; 100 StunMessage* request, const rtc::SocketAddress& addr [all...] |
/external/webrtc/talk/app/webrtc/java/jni/ |
native_handle_impl.h | 55 const rtc::Callback0<void>& no_longer_used); 57 rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override; 59 rtc::scoped_refptr<AndroidTextureBuffer> ScaleAndRotate( 72 rtc::Callback0<void> no_longer_used_cb_;
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/external/webrtc/talk/app/webrtc/objc/ |
RTCAudioTrack+Internal.h | 35 rtc::scoped_refptr<webrtc::AudioTrackInterface> audioTrack;
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RTCPeerConnectionFactory+Internal.h | 35 @property(nonatomic, assign) rtc::scoped_refptr<
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RTCVideoSource+Internal.h | 35 rtc::scoped_refptr<webrtc::VideoSourceInterface>videoSource;
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RTCVideoTrack+Internal.h | 38 rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
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/external/webrtc/talk/app/webrtc/ |
videotrackrenderers.cc | 43 rtc::CritScope cs(&critical_section_); 48 rtc::CritScope cs(&critical_section_); 53 rtc::CritScope cs(&critical_section_); 58 rtc::CritScope cs(&critical_section_);
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/external/webrtc/talk/session/media/ |
mediamonitor.h | 42 class MediaMonitor : public rtc::MessageHandler, 45 MediaMonitor(rtc::Thread* worker_thread, 46 rtc::Thread* monitor_thread); 53 void OnMessage(rtc::Message *message); 58 rtc::CriticalSection crit_; 59 rtc::Thread* worker_thread_; 60 rtc::Thread* monitor_thread_; 69 MediaMonitorT(MC* media_channel, rtc::Thread* worker_thread, 70 rtc::Thread* monitor_thread)
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
neteq_quality_test.h | 61 rtc::scoped_ptr<UniformLoss> uniform_loss_model_; 122 rtc::scoped_ptr<InputAudioFile> in_file_; 123 rtc::scoped_ptr<AudioSink> output_; 126 rtc::scoped_ptr<RtpGenerator> rtp_generator_; 127 rtc::scoped_ptr<NetEq> neteq_; 128 rtc::scoped_ptr<LossModel> loss_model_; 130 rtc::scoped_ptr<int16_t[]> in_data_; 131 rtc::scoped_ptr<uint8_t[]> payload_; 132 rtc::scoped_ptr<int16_t[]> out_data_;
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