| /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_);
|
| 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_);
|
| /external/webrtc/webrtc/base/ |
| 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
|
| asyncpacketsocket.cc | 13 namespace rtc { namespace
29 }; // namespace rtc
|
| 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
|
| 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
|
| proxyinfo.cc | 13 namespace rtc { namespace
24 } // namespace rtc
|
| socketserver.h | 16 namespace rtc { namespace
44 } // namespace rtc
|
| sslfingerprint.h | 20 namespace rtc { namespace
26 const rtc::SSLIdentity* identity);
29 const rtc::SSLCertificate* cert);
47 rtc::Buffer digest;
50 } // namespace rtc
|
| thread_checker_impl.h | 19 namespace rtc { namespace
46 } // namespace rtc
|
| versionparsing.h | 16 namespace rtc { namespace
33 } // namespace rtc
|
| /external/webrtc/webrtc/libjingle/xmpp/ |
| xmppthread.h | 23 public rtc::Thread, buzz::XmppPumpNotify, rtc::MessageHandler {
39 void OnMessage(rtc::Message* pmsg);
|
| 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_;
|
| /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;
|
| 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;
|
| /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_;
|
| /external/webrtc/talk/app/webrtc/objc/ |
| RTCAudioTrack+Internal.h | 35 rtc::scoped_refptr<webrtc::AudioTrackInterface> audioTrack;
|
| RTCPeerConnectionFactory+Internal.h | 35 @property(nonatomic, assign) rtc::scoped_refptr<
|
| RTCVideoSource+Internal.h | 35 rtc::scoped_refptr<webrtc::VideoSourceInterface>videoSource;
|
| RTCVideoTrack+Internal.h | 38 rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
|
| /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_);
|
| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| transient_detector.h | 58 rtc::scoped_ptr<WPDTree> wpd_tree_;
62 rtc::scoped_ptr<MovingMoments> moving_moments_[kLeaves];
64 rtc::scoped_ptr<float[]> first_moments_;
65 rtc::scoped_ptr<float[]> second_moments_;
|
| /external/webrtc/webrtc/sound/ |
| linuxsoundsystem.h | 17 namespace rtc { namespace
40 } // namespace rtc
|
| nullsoundsystemfactory.cc | 15 namespace rtc { namespace
32 } // namespace rtc
|
