/external/chromium_org/third_party/webrtc/base/ |
H A D | nattypes.h | 14 namespace rtc { namespace 45 } // namespace rtc
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H A D | platform_file.h | 20 namespace rtc { namespace 42 } // namespace rtc
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H A D | ratelimiter.cc | 13 namespace rtc { namespace 29 } // namespace rtc
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H A D | ratetracker.h | 17 namespace rtc { namespace 40 } // namespace rtc
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H A D | scoped_autorelease_pool.h | 34 namespace rtc { namespace 56 } // namespace rtc
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H A D | sha1.h | 8 // Ported to C++, Google style, under namespace rtc and uses basictypes.h 15 namespace rtc { namespace 32 } // namespace rtc
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H A D | socketaddresspair.h | 16 namespace rtc { namespace 39 } // namespace rtc
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H A D | socketfactory.h | 17 namespace rtc { namespace 36 } // namespace rtc
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H A D | socketserver.h | 16 namespace rtc { namespace 42 } // namespace rtc
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H A D | thread_checker.h | 33 namespace rtc { namespace 89 } // namespace rtc
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H A D | thread_checker_impl.cc | 17 namespace rtc { namespace 44 } // namespace rtc
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H A D | thread_checker_impl.h | 18 namespace rtc { namespace 49 } // namespace rtc
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H A D | timing.h | 18 namespace rtc { namespace 58 } // namespace rtc
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H A D | urlencode_unittest.cc | 16 using rtc::UrlEncode; 66 ASSERT_EQ(7, rtc::UrlEncodeWithoutEncodingSpaceAsPlus(source, dest, 74 ASSERT_EQ(3, rtc::UrlDecode(source, dest)); 81 ASSERT_EQ(3, rtc::UrlDecodeWithoutEncodingSpaceAsPlus(source, dest));
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H A D | win32socketinit.cc | 15 namespace rtc { namespace 46 } // namespace rtc
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | network_predictor.h | 41 scoped_ptr<rtc::ExpFilter> loss_rate_filter_;
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
H A D | peerconnectionendtoend_unittest.cc | 71 rtc::replace_substrs(line.c_str(), line.length(), 78 rtc::replace_substrs(line.c_str(), line.length(), 129 typedef std::vector<rtc::scoped_refptr<DataChannelInterface> > 133 : caller_(new rtc::RefCountedObject<PeerConnectionTestWrapper>( 135 callee_(new rtc::RefCountedObject<PeerConnectionTestWrapper>( 137 rtc::InitializeSSL(NULL); 225 rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc1_observer( 228 rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc2_observer( 266 rtc::CleanupSSL(); 270 rtc [all...] |
H A D | streamcollection.h | 41 static rtc::scoped_refptr<StreamCollection> Create() { 42 rtc::RefCountedObject<StreamCollection>* implementation = 43 new rtc::RefCountedObject<StreamCollection>(); 47 static rtc::scoped_refptr<StreamCollection> Create( 49 rtc::RefCountedObject<StreamCollection>* implementation = 50 new rtc::RefCountedObject<StreamCollection>(streams); 118 typedef std::vector<rtc::scoped_refptr<MediaStreamInterface> >
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H A D | sctputils.cc | 51 bool ParseDataChannelOpenMessage(const rtc::Buffer& payload, 57 rtc::ByteBuffer buffer(payload.data(), payload.length()); 128 bool ParseDataChannelOpenAckMessage(const rtc::Buffer& payload) { 129 rtc::ByteBuffer buffer(payload.data(), payload.length()); 146 rtc::Buffer* payload) { 174 rtc::ByteBuffer buffer( 176 rtc::ByteBuffer::ORDER_NETWORK); 189 void WriteDataChannelOpenAckMessage(rtc::Buffer* payload) { 190 rtc::ByteBuffer buffer(rtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
H A D | peerconnectiontestwrapper.h | 55 rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel( 109 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( 114 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> 116 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; 117 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> 119 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; 120 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
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/external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
H A D | dtlstransportchannel_unittest.cc | 45 if (!(rtc::SSLStreamAdapter::feature())) { \ 67 rtc::Thread* signaling_thread, 68 rtc::Thread* worker_thread) : 83 identity_.reset(rtc::SSLIdentity::Generate(name_)); 85 rtc::SSLIdentity* identity() { return identity_.get(); } 138 void Negotiate(rtc::SSLIdentity* local_identity, 139 rtc::SSLIdentity* remote_identity, 144 rtc::scoped_ptr<rtc::SSLFingerprint> local_fingerprint; 145 rtc [all...] |
H A D | transportdescriptionfactory.cc | 50 rtc::scoped_ptr<TransportDescription> desc(new TransportDescription()); 64 desc->ice_ufrag = rtc::CreateRandomString(ICE_UFRAG_LENGTH); 65 desc->ice_pwd = rtc::CreateRandomString(ICE_PWD_LENGTH); 89 rtc::scoped_ptr<TransportDescription> desc(new TransportDescription()); 117 desc->ice_ufrag = rtc::CreateRandomString(ICE_UFRAG_LENGTH); 118 desc->ice_pwd = rtc::CreateRandomString(ICE_PWD_LENGTH); 164 rtc::SSLFingerprint::Create(digest_alg, identity_));
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H A D | stunrequest.cc | 42 StunRequestManager::StunRequestManager(rtc::Thread* thread) 125 rtc::ByteBuffer buf(data, size); 126 rtc::scoped_ptr<StunMessage> response(iter->second->msg_->CreateNew()); 137 rtc::CreateRandomString(kStunTransactionIdLength)); 144 rtc::CreateRandomString(kStunTransactionIdLength)); 173 return rtc::TimeSince(tstamp_); 182 void StunRequest::OnMessage(rtc::Message* pmsg) { 192 tstamp_ = rtc::Time(); 194 rtc::ByteBuffer buf; 203 int delay = DELAY_UNIT * rtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | webrtcpassthroughrender.cc | 48 rtc::CritScope cs(&stream_critical_); 56 rtc::CritScope cs(&stream_critical_); 62 rtc::CritScope cs(&stream_critical_); 68 rtc::CritScope cs(&stream_critical_); 76 rtc::CriticalSection stream_critical_; 97 rtc::CritScope cs(&render_critical_); 113 rtc::CritScope cs(&render_critical_); 127 rtc::CritScope cs(&render_critical_); 146 rtc::CritScope cs(&render_critical_); 156 rtc [all...] |
/external/chromium_org/content/renderer/p2p/ |
H A D | ipc_network_manager_unittest.cc | 54 std::vector<rtc::Network*> networks; 55 rtc::IPAddress ip_address; 104 EXPECT_TRUE(rtc::IPFromString(kIPv6PublicAddrString1, &ip_address)); 106 EXPECT_TRUE(rtc::IPFromString(kIPv6PublicAddrString2, &ip_address)); 111 EXPECT_TRUE(rtc::IPFromString(kIPv6PublicAddrString2, &ip_address));
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