| /external/chromium_org/third_party/webrtc/base/ |
| 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...] |
| asyncfile.h | 14 #include "webrtc/base/sigslot.h"
33 sigslot::signal1<AsyncFile*> SignalReadEvent;
34 sigslot::signal1<AsyncFile*> SignalWriteEvent;
35 sigslot::signal2<AsyncFile*, int> SignalCloseEvent;
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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;
|
| 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
[all...] |
| 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<> {
|
| 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;
|
| asyncresolverinterface.h | 14 #include "webrtc/base/sigslot.h"
42 sigslot::signal1<AsyncResolverInterface*> SignalDone;
|
| cpumonitor.h | 17 #include "webrtc/base/sigslot.h"
95 : public rtc::MessageHandler, public sigslot::has_slots<> {
104 sigslot::signal4<int, int, float, float> SignalUpdate;
|
| proxyserver.h | 31 class ProxyBinding : public sigslot::has_slots<> {
34 sigslot::signal1<ProxyBinding*> SignalDestroyed;
60 class ProxyServer : public sigslot::has_slots<> {
|
| socketpool.h | 17 #include "webrtc/base/sigslot.h"
46 class StreamCache : public StreamPool, public sigslot::has_slots<> {
95 class ReuseSocketPool : public StreamPool, public sigslot::has_slots<> {
|
| /external/chromium_org/third_party/libjingle/source/talk/xmpp/ |
| 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...] |
| 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;
|
| 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*,
|
| 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/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;
|
| call.h | 70 class AudioSourceProxy: public AudioSourceContext, public sigslot::has_slots<> {
83 class Call : public rtc::MessageHandler, public sigslot::has_slots<> {
161 sigslot::signal0<> SignalSetupToCallVoicemail;
162 sigslot::signal2<Call*, Session*> SignalAddSession;
163 sigslot::signal2<Call*, Session*> SignalRemoveSession;
164 sigslot::signal3<Call*, Session*, Session::State>
166 sigslot::signal3<Call*, Session*, Session::Error>
168 sigslot::signal3<Call*, Session*, const std::string &>
170 sigslot::signal2<Call*, const std::vector<ConnectionInfo> &>
172 sigslot::signal2<Call*, const VoiceMediaInfo&> SignalMediaMonitor
[all...] |
| audiomonitor.h | 33 #include "webrtc/base/sigslot.h"
48 public sigslot::has_slots<> {
59 sigslot::signal2<AudioMonitor*, const AudioInfo&> SignalUpdate;
|
| channel.h | 49 #include "webrtc/base/sigslot.h"
75 : public rtc::MessageHandler, public sigslot::has_slots<>,
163 void UnregisterSendSink(sigslot::has_slots<>* sink,
196 void UnregisterRecvSink(sigslot::has_slots<>* sink,
225 sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
228 sigslot::repeater2<BaseChannel*, bool> SignalAutoMuted;
361 sigslot::signal3<const void*, size_t, bool> SignalSendPacketPreCrypto;
362 sigslot::signal3<const void*, size_t, bool> SignalSendPacketPostCrypto;
363 sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPreCrypto;
364 sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPostCrypto
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/ |
| LICENSE_THIRD_PARTY | 25 Governed by http://sigslot.sourceforge.net/#license (Public domain):
26 base/sigslot.h
|
| /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;
|
| portallocator.h | 37 #include "webrtc/base/sigslot.h"
78 class PortAllocatorSession : public sigslot::has_slots<> {
101 sigslot::signal2<PortAllocatorSession*, PortInterface*> SignalPortReady;
102 sigslot::signal2<PortAllocatorSession*,
104 sigslot::signal1<PortAllocatorSession*> SignalCandidatesAllocationDone;
108 sigslot::signal1<PortAllocatorSession*> SignalDestroyed;
124 class PortAllocator : public sigslot::has_slots<> {
|
| /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*,
|
