| /external/webrtc/talk/app/webrtc/ |
| D | fakemediacontroller.h | 40 : channel_manager_(channel_manager), call_(call) { in FakeMediaController()
42 RTC_DCHECK(nullptr != call_); in FakeMediaController()
45 webrtc::Call* call_w() override { return call_; } in call_w()
52 webrtc::Call* call_; variable
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| D | mediacontroller.cc | 58 return call_.get(); in call_w()
74 call_.reset(webrtc::Call::Create(config)); in Construct_w()
78 call_.reset(); in Destruct_w()
83 rtc::scoped_ptr<webrtc::Call> call_; member in __anonda9d8f400111::MediaController
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| /external/v8/src/compiler/ |
| D | js-inlining.cc | 37 explicit JSCallAccessor(Node* call) : call_(call) { in JSCallAccessor()
44 return call_->InputAt(0); in target()
48 DCHECK_EQ(IrOpcode::kJSCallFunction, call_->opcode()); in receiver()
49 return call_->InputAt(1); in receiver()
53 DCHECK_EQ(IrOpcode::kJSCallConstruct, call_->opcode()); in new_target()
54 return call_->InputAt(formal_arguments() + 1); in new_target()
58 return NodeProperties::GetFrameStateInput(call_, 1); in frame_state_before()
63 return NodeProperties::GetFrameStateInput(call_, 0); in frame_state_after()
70 return call_->op()->ValueInputCount() - 2; in formal_arguments()
74 Node* call_; member in v8::internal::compiler::JSCallAccessor
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| /external/webrtc/webrtc/call/ |
| D | call_unittest.cc | 27 call_.reset(webrtc::Call::Create(config)); in CallHelper()
30 webrtc::Call* operator->() { return call_.get(); } in operator ->()
34 rtc::scoped_ptr<webrtc::Call> call_; member
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| /external/webrtc/talk/media/webrtc/ |
| D | webrtcvoiceengine_unittest.cc | 78 : call_(webrtc::Call::Config()), in WebRtcVoiceEngineTestFake()
88 channel_ = engine_.CreateChannel(&call_, cricket::AudioOptions()); in SetupEngine()
108 EXPECT_TRUE(call_.GetAudioSendStream(kSsrc1)); in SetupForMultiSendStream()
111 EXPECT_FALSE(call_.GetAudioSendStream(kSsrc1)); in SetupForMultiSendStream()
123 const auto* send_stream = call_.GetAudioSendStream(ssrc); in GetSendStream()
129 const auto* send_stream = call_.GetAudioSendStream(ssrc); in GetSendStreamConfig()
135 const auto* recv_stream = call_.GetAudioReceiveStream(ssrc); in GetRecvStreamConfig()
142 channel_ = engine_.CreateChannel(&call_, cricket::AudioOptions()); in TestInsertDtmf()
229 EXPECT_NE(call_.GetAudioSendStream(kSsrc1), in TestSetSendRtpHeaderExtensions()
230 call_.GetAudioSendStream(kSsrc2)); in TestSetSendRtpHeaderExtensions()
[all …]
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| D | webrtcvideoengine2.cc | 620 : call_(call), in WebRtcVideoChannel2()
878 call_->SetBitrateConfig(bitrate_config_); in SetSendCodecs()
972 call_, sp, config, external_encoder_factory_, options_, in AddSendStream()
1101 call_, sp, config, external_decoder_factory_, default_stream, in AddRecvStream()
1207 webrtc::Call::Stats stats = call_->GetStats(); in GetStats()
1298 call_->Receiver()->DeliverPacket( in OnPacketReceived()
1341 if (call_->Receiver()->DeliverPacket( in OnPacketReceived()
1359 call_->Receiver()->DeliverPacket( in OnRtcpReceived()
1367 call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown); in OnReadyToSend()
1462 call_->SetBitrateConfig(bitrate_config_); in SetMaxSendBandwidth()
[all …]
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| D | webrtcvoiceengine.cc | 1082 call_(call), in WebRtcAudioSendStream()
1098 call_->DestroyAudioSendStream(stream_); in ~WebRtcAudioSendStream()
1105 call_->DestroyAudioSendStream(stream_); in RecreateAudioSendStream()
1110 stream_ = call_->CreateAudioSendStream(config_); in RecreateAudioSendStream()
1190 webrtc::Call* call_ = nullptr; member in cricket::WebRtcVoiceMediaChannel::WebRtcAudioSendStream
1210 : call_(call), in WebRtcAudioReceiveStream()
1223 call_->DestroyAudioReceiveStream(stream_); in ~WebRtcAudioReceiveStream()
1257 call_->DestroyAudioReceiveStream(stream_); in RecreateAudioReceiveStream()
1263 stream_ = call_->CreateAudioReceiveStream(config_); in RecreateAudioReceiveStream()
1268 webrtc::Call* call_ = nullptr; member in cricket::WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream
[all …]
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| D | webrtcvideoengine2.h | 358 webrtc::Call* const call_; variable
444 webrtc::Call* const call_; variable
498 webrtc::Call* const call_; variable
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| D | webrtcvoiceengine.h | 266 webrtc::Call* const call_ = nullptr; variable
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| D | webrtcvideoengine2_unittest.cc | 120 call_(webrtc::Call::Create(webrtc::Call::Config())), in WebRtcVideoEngine2Test()
157 rtc::scoped_ptr<webrtc::Call> call_; member in cricket::WebRtcVideoEngine2Test
345 engine_.CreateChannel(call_.get(), cricket::VideoOptions())); in TEST_F()
358 engine_.CreateChannel(call_.get(), cricket::VideoOptions())); in TEST_F()
421 call_.reset(fake_call); in TEST_F()
481 call_.reset(fake_call); in TEST_F()
537 engine_.CreateChannel(call_.get(), cricket::VideoOptions()); in SetUpForExternalEncoderFactory()
552 engine_.CreateChannel(call_.get(), cricket::VideoOptions()); in SetUpForExternalDecoderFactory()
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| /external/webrtc/talk/media/base/ |
| D | videoengine_unittest.h | 134 : call_(webrtc::Call::Create(webrtc::Call::Config())) {} in call_() function
146 engine_.CreateChannel(call_.get(), cricket::VideoOptions())); in SetUp()
1485 const rtc::scoped_ptr<webrtc::Call> call_; variable
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| /external/webrtc/webrtc/video/ |
| D | video_send_stream_tests.cc | 1736 call_ = sender_call; in TEST_F()
1749 call_->SetBitrateConfig(bitrate_config); in TEST_F()
1767 webrtc::Call* call_; in TEST_F() member in webrtc::TEST_F::EncoderBitrateThresholdObserver
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