| /external/webrtc/webrtc/base/ |
| asyncfile.h | 14 #include "webrtc/base/sigslot.h"
33 sigslot::signal1<AsyncFile*> SignalReadEvent;
34 sigslot::signal1<AsyncFile*> SignalWriteEvent;
35 sigslot::signal2<AsyncFile*, int> SignalCloseEvent;
|
| sigslot_unittest.cc | 11 #include "webrtc/base/sigslot.h"
17 static bool TemplateIsST(const sigslot::single_threaded* p) {
22 static bool TemplateIsMT(const sigslot::multi_threaded_local* p) {
26 class SigslotDefault : public testing::Test, public sigslot::has_slots<> {
28 sigslot::signal0<> signal_;
31 template<class slot_policy = sigslot::single_threaded,
32 class signal_policy = sigslot::single_threaded>
33 class SigslotReceiver : public sigslot::has_slots<slot_policy> {
40 void Connect(sigslot::signal0<signal_policy>* signal) {
58 sigslot::signal0<signal_policy>* signal_
[all...] |
| asyncpacketsocket.h | 15 #include "webrtc/base/sigslot.h"
67 class AsyncPacketSocket : public sigslot::has_slots<> {
109 sigslot::signal5<AsyncPacketSocket*, const char*, size_t,
114 sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket;
117 sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend;
123 sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady;
127 sigslot::signal1<AsyncPacketSocket*> SignalConnect;
131 sigslot::signal2<AsyncPacketSocket*, int> SignalClose;
134 sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection;
|
| networkmonitor.h | 16 #include "webrtc/base/sigslot.h"
44 sigslot::signal0<> SignalNetworksChanged;
56 public sigslot::has_slots<> {
|
| asyncsocket.h | 15 #include "webrtc/base/sigslot.h"
35 sigslot::signal1<AsyncSocket*,
36 sigslot::multi_threaded_local> SignalReadEvent;
38 sigslot::signal1<AsyncSocket*,
39 sigslot::multi_threaded_local> SignalWriteEvent;
40 sigslot::signal1<AsyncSocket*> SignalConnectEvent; // connected
41 sigslot::signal2<AsyncSocket*, int> SignalCloseEvent; // closed
44 class AsyncSocketAdapter : public AsyncSocket, public sigslot::has_slots<> {
|
| httpserver.h | 44 // Due to sigslot issues, we can't destroy some streams at an arbitrary time.
45 sigslot::signal3<HttpServer*, int, StreamInterface*> SignalConnectionClosed;
54 sigslot::signal3<HttpServer*, HttpServerTransaction*, bool*>
62 sigslot::signal2<HttpServer*, HttpServerTransaction*> SignalHttpRequest;
66 sigslot::signal3<HttpServer*, HttpServerTransaction*, int>
77 sigslot::signal1<HttpServer*> SignalCloseAllComplete;
116 class HttpListenServer : public HttpServer, public sigslot::has_slots<> {
|
| asyncresolverinterface.h | 14 #include "webrtc/base/sigslot.h"
42 sigslot::signal1<AsyncResolverInterface*> SignalDone;
|
| sigslottester.h | 30 // sigslot::signal1<const std::string&> foo;
43 #include "webrtc/base/sigslot.h"
54 class SigslotTester1 : public sigslot::has_slots<> {
56 SigslotTester1(sigslot::signal1<A1>* signal,
78 class SigslotTester2 : public sigslot::has_slots<> {
80 SigslotTester2(sigslot::signal2<A1, A2>* signal,
104 class SigslotTester3 : public sigslot::has_slots<> {
106 SigslotTester3(sigslot::signal3<A1, A2, A3>* signal,
133 class SigslotTester4 : public sigslot::has_slots<> {
135 SigslotTester4(sigslot::signal4<A1, A2, A3, A4>* signal
[all...] |
| sigslottester_unittest.cc | 16 #include "webrtc/base/sigslot.h"
21 sigslot::signal1<int> source1;
36 sigslot::signal2<int, char> source2;
56 sigslot::signal1<const std::string&> source1;
66 sigslot::signal1<const std::string*> source1;
77 sigslot::signal1<std::string* const> source1;
|
| /external/webrtc/talk/app/webrtc/ |
| mediastreamobserver.h | 33 #include "webrtc/base/sigslot.h"
48 sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
50 sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
52 sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
54 sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
|
| /external/webrtc/talk/session/media/ |
| currentspeakermonitor.h | 37 #include "webrtc/base/sigslot.h"
46 sigslot::signal2<AudioSourceContext*, const cricket::AudioInfo&>
48 sigslot::signal1<AudioSourceContext*> SignalMediaStreamsReset;
49 sigslot::signal3<AudioSourceContext*,
63 class CurrentSpeakerMonitor : public sigslot::has_slots<> {
79 sigslot::signal2<CurrentSpeakerMonitor*, uint32_t> SignalUpdate;
|
| audiomonitor.h | 33 #include "webrtc/base/sigslot.h"
48 public sigslot::has_slots<> {
59 sigslot::signal2<AudioMonitor*, const AudioInfo&> SignalUpdate;
|
| /external/webrtc/webrtc/libjingle/xmpp/ |
| asyncsocket.h | 16 #include "webrtc/base/sigslot.h"
63 sigslot::signal0<> SignalConnected;
64 sigslot::signal0<> SignalSSLConnected;
65 sigslot::signal0<> SignalClosed;
66 sigslot::signal0<> SignalRead;
67 sigslot::signal0<> SignalError;
|
| hangoutpubsubclient.h | 22 #include "webrtc/base/sigslot.h"
37 class HangoutPubSubClient : public sigslot::has_slots<> {
50 sigslot::signal3<const std::string&, bool, bool> SignalPresenterStateChange;
52 sigslot::signal3<const std::string&, bool, bool> SignalAudioMuteStateChange;
54 sigslot::signal3<const std::string&, bool, bool> SignalVideoMuteStateChange;
56 sigslot::signal3<const std::string&, bool, bool> SignalVideoPauseStateChange;
58 sigslot::signal3<const std::string&, bool, bool> SignalRecordingStateChange;
60 sigslot::signal3<const std::string&,
64 sigslot::signal2<const std::string&, const std::string&> SignalMediaBlock;
67 sigslot::signal2<const std::string&, const XmlElement*> SignalRequestError
[all...] |
| pubsubclient.h | 19 #include "webrtc/base/sigslot.h"
36 class PubSubClient : public sigslot::has_slots<> {
53 sigslot::signal2<PubSubClient*,
56 sigslot::signal2<PubSubClient*,
59 sigslot::signal4<PubSubClient*,
64 sigslot::signal3<PubSubClient*,
68 sigslot::signal3<PubSubClient*,
72 sigslot::signal2<PubSubClient*,
|
| mucroomuniquehangoutidtask.h | 26 sigslot::signal2<MucRoomUniqueHangoutIdTask*, const std::string&> SignalResult;
|
| presencereceivetask.h | 14 #include "webrtc/base/sigslot.h"
36 sigslot::signal1<const PresenceStatus&> PresenceUpdate;
|
| pubsubtasks.h | 18 #include "webrtc/base/sigslot.h"
38 sigslot::signal2<PubSubRequestTask*,
58 sigslot::signal2<PubSubReceiveTask*,
83 sigslot::signal1<PubSubPublishTask*> SignalResult;
103 sigslot::signal1<PubSubRetractTask*> SignalResult;
|
| /external/webrtc/webrtc/p2p/base/ |
| transportchannel.h | 23 #include "webrtc/base/sigslot.h"
51 class TransportChannel : public sigslot::has_slots<> {
78 sigslot::signal1<TransportChannel*> SignalWritableState;
80 sigslot::signal1<TransportChannel*> SignalReadyToSend;
81 sigslot::signal1<TransportChannel*> SignalReceivingState;
84 sigslot::signal2<TransportChannel*, DtlsTransportState> SignalDtlsState;
143 sigslot::signal5<TransportChannel*, const char*,
147 sigslot::signal2<TransportChannel*, const rtc::SentPacket&> SignalSentPacket;
152 sigslot::signal2<TransportChannel*, const Candidate&> SignalRouteChange;
155 sigslot::signal1<TransportChannel*> SignalDestroyed
[all...] |
| transportchannelimpl.h | 71 sigslot::signal1<TransportChannelImpl*> SignalGatheringState;
81 sigslot::signal2<TransportChannelImpl*, const Candidate&>
100 sigslot::signal1<TransportChannelImpl*> SignalRoleConflict;
104 sigslot::signal1<TransportChannelImpl*> SignalConnectionRemoved;
|
| portinterface.h | 90 sigslot::signal6<PortInterface*, const rtc::SocketAddress&,
105 sigslot::signal1<PortInterface*> SignalDestroyed;
108 sigslot::signal1<PortInterface*> SignalRoleConflict;
115 sigslot::signal4<PortInterface*, const char*, size_t,
119 sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
|
| /external/webrtc/webrtc/sound/ |
| soundinputstreaminterface.h | 15 #include "webrtc/base/sigslot.h"
56 sigslot::signal3<const void *, size_t,
|
| soundoutputstreaminterface.h | 15 #include "webrtc/base/sigslot.h"
61 sigslot::signal2<size_t, SoundOutputStreamInterface *> SignalBufferSpace;
|
| /external/webrtc/talk/app/webrtc/test/ |
| peerconnectiontestwrapper.h | 35 #include "webrtc/base/sigslot.h"
40 public sigslot::has_slots<> {
89 sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
90 sigslot::signal3<const std::string&,
93 sigslot::signal1<std::string*> SignalOnSdpCreated;
94 sigslot::signal1<const std::string&> SignalOnSdpReady;
95 sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
|
| /external/webrtc/webrtc/p2p/client/ |
| socketmonitor.h | 17 #include "webrtc/base/sigslot.h"
33 public sigslot::has_slots<> {
43 sigslot::signal2<ConnectionMonitor*,
|
