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  /external/chromium_org/third_party/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"
68 class AsyncPacketSocket : public sigslot::has_slots<> {
110 sigslot::signal5<AsyncPacketSocket*, const char*, size_t,
115 sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend;
121 sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady;
125 sigslot::signal1<AsyncPacketSocket*> SignalConnect;
129 sigslot::signal2<AsyncPacketSocket*, int> SignalClose;
132 sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection;
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;
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<> {
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
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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/chromium_org/third_party/libjingle/source/talk/media/base/
videoprocessor.h 32 #include "webrtc/base/sigslot.h"
36 class VideoProcessor : public sigslot::has_slots<> {
voiceprocessor.h 33 #include "webrtc/base/sigslot.h"
44 class VoiceProcessor : public sigslot::has_slots<> {
  /external/chromium_org/third_party/libjingle/source/talk/xmpp/
asyncsocket.h 33 #include "webrtc/base/sigslot.h"
80 sigslot::signal0<> SignalConnected;
81 sigslot::signal0<> SignalSSLConnected;
82 sigslot::signal0<> SignalClosed;
83 sigslot::signal0<> SignalRead;
84 sigslot::signal0<> SignalError;
hangoutpubsubclient.h 39 #include "webrtc/base/sigslot.h"
54 class HangoutPubSubClient : public sigslot::has_slots<> {
67 sigslot::signal3<const std::string&, bool, bool> SignalPresenterStateChange;
69 sigslot::signal3<const std::string&, bool, bool> SignalAudioMuteStateChange;
71 sigslot::signal3<const std::string&, bool, bool> SignalVideoMuteStateChange;
73 sigslot::signal3<const std::string&, bool, bool> SignalVideoPauseStateChange;
75 sigslot::signal3<const std::string&, bool, bool> SignalRecordingStateChange;
77 sigslot::signal3<const std::string&,
81 sigslot::signal2<const std::string&, const std::string&> SignalMediaBlock;
84 sigslot::signal2<const std::string&, const XmlElement*> SignalRequestError
    [all...]
pubsubclient.h 36 #include "webrtc/base/sigslot.h"
53 class PubSubClient : public sigslot::has_slots<> {
70 sigslot::signal2<PubSubClient*,
73 sigslot::signal2<PubSubClient*,
76 sigslot::signal4<PubSubClient*,
81 sigslot::signal3<PubSubClient*,
85 sigslot::signal3<PubSubClient*,
89 sigslot::signal2<PubSubClient*,
mucroomuniquehangoutidtask.h 19 sigslot::signal2<MucRoomUniqueHangoutIdTask*, const std::string&> SignalResult;
presencereceivetask.h 31 #include "webrtc/base/sigslot.h"
53 sigslot::signal1<const PresenceStatus&> PresenceUpdate;
pubsubtasks.h 35 #include "webrtc/base/sigslot.h"
55 sigslot::signal2<PubSubRequestTask*,
75 sigslot::signal2<PubSubReceiveTask*,
100 sigslot::signal1<PubSubPublishTask*> SignalResult;
120 sigslot::signal1<PubSubRetractTask*> SignalResult;
  /external/chromium_org/third_party/libjingle/source/talk/examples/call/
presencepushtask.h 37 #include "webrtc/base/sigslot.h"
48 sigslot::signal1<const PresenceStatus&> SignalStatusUpdate;
49 sigslot::signal1<const Jid&> SignalMucJoined;
50 sigslot::signal2<const Jid&, int> SignalMucLeft;
51 sigslot::signal2<const Jid&, const MucPresenceStatus&> SignalMucStatusUpdate;
  /external/chromium_org/third_party/libjingle/source/talk/p2p/base/
transportchannel.h 40 #include "webrtc/base/sigslot.h"
58 class TransportChannel : public sigslot::has_slots<> {
78 sigslot::signal1<TransportChannel*> SignalReadableState;
79 sigslot::signal1<TransportChannel*> SignalWritableState;
81 sigslot::signal1<TransportChannel*> SignalReadyToSend;
126 sigslot::signal5<TransportChannel*, const char*,
132 sigslot::signal2<TransportChannel*, const Candidate&> SignalRouteChange;
135 sigslot::signal1<TransportChannel*> SignalDestroyed;
transportchannelimpl.h 82 sigslot::signal1<TransportChannelImpl*> SignalRequestSignaling;
93 sigslot::signal2<TransportChannelImpl*,
112 sigslot::signal1<TransportChannelImpl*> SignalCandidatesAllocationDone;
116 sigslot::signal1<TransportChannelImpl*> SignalRoleConflict;
120 sigslot::signal1<TransportChannelImpl*> SignalConnectionRemoved;
  /external/chromium_org/third_party/libjingle/source/talk/session/media/
currentspeakermonitor.h 37 #include "webrtc/base/sigslot.h"
48 sigslot::signal2<AudioSourceContext*, const cricket::AudioInfo&>
50 sigslot::signal2<AudioSourceContext*, cricket::BaseSession*>
52 sigslot::signal4<AudioSourceContext*, cricket::BaseSession*,
66 class CurrentSpeakerMonitor : public sigslot::has_slots<> {
85 sigslot::signal2<CurrentSpeakerMonitor*, uint32> SignalUpdate;
typingmonitor.h 60 : public rtc::MessageHandler, public sigslot::has_slots<> {
66 sigslot::signal2<BaseChannel*, bool> SignalMuted;
  /external/chromium_org/third_party/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/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/
peerconnectiontestwrapper.h 35 #include "webrtc/base/sigslot.h"
45 public sigslot::has_slots<> {
95 sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
96 sigslot::signal3<const std::string&,
99 sigslot::signal1<std::string*> SignalOnSdpCreated;
100 sigslot::signal1<const std::string&> SignalOnSdpReady;
101 sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
  /external/chromium_org/third_party/libjingle/source/talk/p2p/client/
socketmonitor.h 35 #include "webrtc/base/sigslot.h"
41 public sigslot::has_slots<> {
53 sigslot::signal2<SocketMonitor*,

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