/external/v8/src/snapshot/ |
D | serialize.cc | 1268 sink_(sink), in Serializer() 1353 sink_(sink), in ObjectSerializer() 1399 SnapshotByteSink* sink_; member in v8::internal::Serializer::ObjectSerializer 1410 ObjectSerializer obj_serializer(this, obj, sink_, kPlain, kStartOfObject); in SerializeDeferredObjects() 1413 sink_->Put(kSynchronize, "Finished with deferred objects"); in SerializeDeferredObjects() 1439 sink_->Put(kSkip, "Skip"); in VisitPointers() 1440 sink_->PutInt(kPointerSize, "SkipOneWord"); in VisitPointers() 1442 sink_->Put(kOnePointerRawData, "Smi"); in VisitPointers() 1444 sink_->Put(reinterpret_cast<byte*>(current)[i], "Byte"); in VisitPointers() 1486 sink_->Put(kOnePointerRawData, "Smi"); in VisitPointers() [all …]
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D | serialize.h | 523 sink_->Put(kSkip, "SkipFromSerializeObject"); in FlushSkip() 524 sink_->PutInt(skip, "SkipDistanceFromSerializeObject"); in FlushSkip() 556 SnapshotByteSink* sink() const { return sink_; } in sink() 567 SnapshotByteSink* sink_; variable
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/external/webrtc/webrtc/base/ |
D | win32socketserver.cc | 231 closing_(false), close_error_(0), sink_(NULL), dns_(NULL) { in Win32Socket() 271 if (sink_) in SetTimeout() 272 ::SetTimer(sink_->handle(), 1, ms, 0); in SetTimeout() 331 if (!sink_) { in Connect() 337 dns->handle = WSAAsyncGetHostByName(sink_->handle(), WM_DNSNOTIFY, in Connect() 506 if (sink_) { in Close() 507 sink_->Dispose(); in Close() 508 sink_ = NULL; in Close() 547 ASSERT(NULL == sink_); in CreateSink() 550 sink_ = new EventSink(this); in CreateSink() [all …]
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D | win32socketserver.h | 81 EventSink * sink_; variable
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/external/webrtc/talk/app/webrtc/ |
D | rtpsender.cc | 36 LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {} in LocalAudioSinkAdapter() 40 if (sink_) in ~LocalAudioSinkAdapter() 41 sink_->OnClose(); in ~LocalAudioSinkAdapter() 50 if (sink_) { in OnData() 51 sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels, in OnData() 58 ASSERT(!sink || !sink_); in SetSink() 59 sink_ = sink; in SetSink()
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D | rtpsender.h | 66 cricket::AudioRenderer::Sink* sink_; variable
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D | rtpsenderreceiver_unittest.cc | 75 sink_ = std::move(sink); in SetRawAudioSink() 79 rtc::scoped_ptr<AudioSinkInterface> sink_; member in webrtc::MockAudioProvider
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D | webrtcsession_unittest.cc | 337 FakeAudioRenderer() : sink_(NULL) {} in FakeAudioRenderer() 339 if (sink_) in ~FakeAudioRenderer() 340 sink_->OnClose(); in ~FakeAudioRenderer() 343 void SetSink(Sink* sink) override { sink_ = sink; } in SetSink() 345 cricket::AudioRenderer::Sink* sink() const { return sink_; } in sink() 347 cricket::AudioRenderer::Sink* sink_; member in FakeAudioRenderer
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/external/webrtc/talk/media/webrtc/ |
D | fakewebrtccall.h | 114 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_; variable
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D | fakewebrtccall.cc | 97 sink_ = std::move(sink); in SetSink()
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/external/webrtc/talk/media/base/ |
D | fakemediaengine.h | 353 sink_ = std::move(sink); in SetRawAudioSink() 428 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_; variable
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