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Searched
refs:sigslot
(Results
1 - 25
of
190
) sorted by null
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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) {
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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<> {
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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/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*,
Completed in 764 milliseconds
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