| /external/chromium_org/third_party/webrtc/video_engine/ |
| D | vie_network_impl.cc | 81 if (vie_channel->Sending()) { in RegisterSendTransport()
101 if (vie_channel->Sending()) { in DeregisterSendTransport()
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| D | vie_channel.cc | 268 if (rtp_rtcp_->Sending() && new_stream) { in SetSendCodec()
325 rtp_rtcp->SetSendingStatus(rtp_rtcp_->Sending()); in SetSendCodec()
917 if (rtp_rtcp_->Sending()) { in SetStartSequenceNumber()
924 assert(!rtp_rtcp_->Sending()); in SetRtpStateForSsrc()
929 assert(!rtp_rtcp_->Sending()); in GetRtpStateForSsrc()
939 if (rtp_rtcp_->Sending()) { in SetRTCPCName()
983 if (!rtp_rtcp_->Sending()) { in SendApplicationDefinedRTCPPacket()
1278 if (rtp_rtcp_->Sending()) { in StartSend()
1304 if (!rtp_rtcp_->Sending()) { in StopSend()
1323 bool ViEChannel::Sending() { in Sending() function in webrtc::ViEChannel
[all …]
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| D | vie_rtp_rtcp_impl.cc | 247 if (vie_channel->Sending()) { in SetStartSequenceNumber()
267 if (vie_channel->Sending()) { in SetRtpStateForSsrc()
329 if (vie_channel->Sending()) { in SetRTCPCName()
369 if (!vie_channel->Sending()) { in SendApplicationDefinedRTCPPacket()
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| D | vie_channel.h | 276 bool Sending();
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| D | test_api.cc | 64 EXPECT_FALSE(module->Sending()); in TEST_F()
66 EXPECT_TRUE(module->Sending()); in TEST_F()
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| /external/chromium_org/third_party/webrtc/voice_engine/ |
| D | transmit_mixer.cc | 310 if (channel->Sending()) { in GetSendCodecInfo()
424 if (channelPtr->Sending()) in DemuxAndMix()
440 if (channel_ptr->Sending()) { in DemuxAndMix()
459 if (channelPtr->Sending()) in EncodeAndSend()
472 if (channel_ptr && channel_ptr->Sending()) in EncodeAndSend()
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| D | shared_data.cc | 84 if (it.GetChannel()->Sending()) in NumOfSendingChannels()
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| D | voe_dtmf_impl.cc | 78 if (!channelPtr->Sending()) in SendTelephoneEvent()
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| D | voe_base_impl.cc | 229 if (channel_ptr->Sending()) { in PushCaptureData()
713 if (channelPtr->Sending()) in StartSend()
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| D | channel.h | 446 bool Sending() const in Sending() function
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| /external/chromium_org/device/serial/ |
| D | serial.mojom | 88 // returning it via |connection|. Sending and receiving data over this
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| /external/clang/test/CodeGenObjC/ |
| D | debug-info-block-line.m | 81 format:@"Sending \"%@\" (%@)", [map name], [map identifier]];
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_rtcp_impl.cc | 175 if (rtcp_sender_.Sending()) { in Process()
450 if (rtcp_sender_.Sending() != sending) { in SetSendingStatus()
477 bool ModuleRtpRtcpImpl::Sending() const { in Sending() function in webrtc::ModuleRtpRtcpImpl
478 return rtcp_sender_.Sending(); in Sending()
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| D | rtcp_sender.h | 82 bool Sending() const;
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| D | rtp_rtcp_impl.h | 108 virtual bool Sending() const OVERRIDE;
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| D | rtp_rtcp.h | 290 virtual bool Sending() const = 0;
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| /external/chromium_org/net/third_party/nss/patches/ |
| D | tlsunique.patch | 31 + /* Sending or receiving a Finished message will set finishedBytes to a
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| /external/pixman/ |
| D | README | 46 Sending patches
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| D | mock_rtp_rtcp.h | 111 MOCK_CONST_METHOD0(Sending,
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| /external/chromium_org/native_client_sdk/src/doc/devguide/coding/ |
| D | message-system.rst | 347 Sending messages back to the JavaScript code
361 Sending and receiving other ``pp::Var`` types
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| /external/mdnsresponder/mDNSCore/ |
| D | DNSCommon.h | 256 #pragma mark - Packet Sending Functions
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| /external/chromium_org/chrome/browser/mac/ |
| D | dock.mm | 136 // Sending a SIGHUP to the Dock seems to be a more reliable way to get the
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| /external/fio/ |
| D | README | 314 your mileage may vary. Sending me patches for other platforms is greatly
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| /external/chromium_org/tools/python/google/httpd_config/ |
| D | mime.types | 4 # given file extension(s). Sending the correct media type to the client
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| /external/iproute2/doc/ |
| D | api-ip6-flowlabels.tex | 87 \section{Sending/receiving flow information.}
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