/external/valgrind/main/gdbserver_tests/ |
H A D | mchelp.stderrB.exp | 0 sending command help to pid .... 2 sending command help debug to pid .... 3 sending command v.kill to pid ....
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H A D | mcinvokeRU.stderrB.exp | 0 sending command v.wait 0 to pid .... 2 sending command v.wait 0 to pid .... 3 sending command v.wait 0 to pid .... 4 sending command v.wait 0 to pid .... 5 sending command v.wait 0 to pid .... 6 sending command v.wait 0 to pid .... 7 sending command v.wait 0 to pid .... 8 sending command v.wait 0 to pid .... 9 sending command v.wait 0 to pid .... 10 sending comman [all...] |
H A D | mcinvokeWS.stderrB.exp | 0 sending command v.wait 0 to pid .... 2 sending command v.wait 0 to pid .... 3 sending command v.wait 0 to pid .... 4 sending command v.wait 0 to pid .... 5 sending command v.wait 0 to pid .... 6 sending command v.wait 0 to pid .... 7 sending command v.wait 0 to pid .... 8 sending command v.wait 0 to pid .... 9 sending command v.wait 0 to pid .... 10 sending comman [all...] |
H A D | send_signal | 6 # the signal sending is done 29 echo sending signal
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H A D | nlvgdbsigqueue.stdoutB.exp | 6 sending signal 7 sending signal
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/external/valgrind/main/none/tests/ |
H A D | pth_blockedsig.stdout.exp | 1 thread CHILD sending SIGUSR1 to thread MAIN
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H A D | pending.stdout.exp | 1 1: sending signal
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/external/chromium_org/third_party/WebKit/Source/devtools/front_end/ |
H A D | networkPanel.css | 140 .network-timing-bar.sending {
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/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | channel_unittest.cc | 550 EXPECT_FALSE(media_channel1_->sending()); 729 // Update the local streams by adding another sending stream. 741 // Update the local streams by removing the first sending stream. 797 // Update the remote streams by adding another sending stream. 872 // Let channel 2 update the content by sending |stream2| and enable SRTP. 884 // Channel 1 replies but stop sending stream1. 899 // Test that we only start playout and sending at the right times. 903 EXPECT_FALSE(media_channel1_->sending()); 905 EXPECT_FALSE(media_channel2_->sending()); 908 EXPECT_FALSE(media_channel1_->sending()); [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | channel.h | 90 sending(false), 98 bool sending; member in struct:webrtc::voe::ChannelState::State 143 state_.sending = enable; 448 return channel_state_.Get().sending;
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H A D | channel.cc | 490 // Note that, the case where the other side stops sending during CNG 742 // This is not needed on sending side, since the codec will 1159 // Ensure that PCMU is used as default codec on the sending side 1311 // This needs to be done before |sending| is set to true. 1315 if (channel_state_.Get().sending) 1325 "StartSend() RTP/RTCP failed to start sending"); 1339 if (!channel_state_.Get().sending) 1353 // Reset sending SSRC and sequence number and triggers direct transmission 1360 "StartSend() RTP/RTCP failed to stop sending"); 3047 if (channel_state_.Get().sending) [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
H A D | rtp_rtcp.h | 259 * Turn on/off sending RTX (RFC 4588). The modes can be set as a combination 264 // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX, 268 // Sets the payload type to use when sending RTX packets. Note that this 273 * Get status of sending RTX (RFC 4588) on a specific SSRC. 281 * sending - on/off 285 virtual int32_t SetSendingStatus(const bool sending) = 0; 295 * sending - on/off 299 virtual int32_t SetSendingMediaStatus(const bool sending) = 0; 459 * Reset RTP data counters for the sending side
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
H A D | mock_rtp_rtcp.h | 110 int32_t(const bool sending)); 114 int32_t(const bool sending));
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_rtcp_impl.cc | 135 // for all outgoing messages sending packets etc. 449 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { argument 450 if (rtcp_sender_.Sending() != sending) { 452 if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) { 460 rtp_sender_.SetSendingStatus(sending); 461 if (sending) { 481 int32_t ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) { argument 482 rtp_sender_.SetSendingMediaStatus(sending); 604 // No RTP sender is interested in sending this packet. 615 // Send padding on one of the modules sending medi [all...] |
H A D | rtp_rtcp_impl.h | 106 virtual int32_t SetSendingStatus(const bool sending) OVERRIDE; 111 virtual int32_t SetSendingMediaStatus(const bool sending) OVERRIDE;
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H A D | rtcp_sender.cc | 214 RTCPSender::SetSendingStatus(const FeedbackState& feedback_state, bool sending) argument 222 if(sending == false && _sending == true) 228 _sending = sending; 1113 // Before sending the TMMBR check the received TMMBN, only an owner is allowed to raise the bitrate
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/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
H A D | videoengine_unittest.h | 580 bool sending = channel_->sending(); local 585 success = SetSend(sending); 764 EXPECT_FALSE(channel_->sending()); 767 EXPECT_FALSE(channel_->sending()); 769 EXPECT_TRUE(channel_->sending()); 773 EXPECT_FALSE(channel_->sending()); 777 EXPECT_FALSE(channel_->sending()); 779 EXPECT_FALSE(channel_->sending()); 1755 // Test sending an [all...] |
H A D | fakemediaengine.h | 51 // A common helper class that handles sending and receiving RTP/RTCP packets. 67 bool sending() const { return sending_; } function in class:cricket::RtpHelper 169 // sending stream. 177 // sending stream.
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | webrtcvideoengine.h | 246 bool sending() const { return sending_; } function in class:cricket::WebRtcVideoMediaChannel 337 // true |ssrc| is the remote ssrc. If |sending| is true the ssrc is 338 // local ssrc. If both |receiving| and |sending| is true the ssrc 373 // Helper function for starting the sending of media on all channels or 377 // Helper function for stop the sending of media on all channels or
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H A D | webrtcvideoengine2.h | 273 bool sending() const { return sending_; } function in class:cricket::WebRtcVideoChannel2
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H A D | webrtcvoiceengine.h | 387 bool sending() const { return send_ != SEND_NOTHING; } function in class:cricket::WebRtcVoiceMediaChannel 465 // send_channels_ contains the channels which are being used for sending. 466 // When the default channel (voe_channel) is used for sending, it is
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H A D | webrtcvideoengine.cc | 556 // TODO(mflodman): Consider sending resolution as part of event, to let 612 void set_sending(bool sending) { sending_ = sending; } argument 613 bool sending() const { return sending_; } function in class:cricket::WebRtcVideoChannelSendInfo 1860 if (send == sending()) { 1865 // We've been asked to start sending. 1875 // We've been asked to stop sending. 1903 // If the default channel is already used for sending create a new channel 1904 // otherwise use the default channel for sending. 2887 // Extension closer to the network, @ socket level before sending 3319 const bool sending = (direction == MD_SEND) || (direction == MD_SENDRECV); local [all...] |
/external/chromium_org/native_client_sdk/src/doc/ |
H A D | faq.rst | 334 Users can opt-in to sending usage statistics and crash information in 338 does so anonymously without sending your application's data or its debug
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/external/iproute2/doc/ |
H A D | ip-tunnels.tex | 227 will route \verb|3FFE::/16| via \verb|sit0|, sending all the packets
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/external/blktrace/doc/ |
H A D | blktrace.tex | 562 start sending requests to the driver. This may happen automatically
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