Searched refs:sending (Results 1 - 25 of 26) sorted by relevance

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/external/valgrind/main/gdbserver_tests/
H A Dmchelp.stderrB.exp0 sending command help to pid ....
2 sending command help debug to pid ....
3 sending command v.kill to pid ....
H A DmcinvokeRU.stderrB.exp0 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
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H A DmcinvokeWS.stderrB.exp0 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
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H A Dsend_signal6 # the signal sending is done
29 echo sending signal
H A Dnlvgdbsigqueue.stdoutB.exp6 sending signal
7 sending signal
/external/valgrind/main/none/tests/
H A Dpth_blockedsig.stdout.exp1 thread CHILD sending SIGUSR1 to thread MAIN
H A Dpending.stdout.exp1 1: sending signal
/external/chromium_org/third_party/WebKit/Source/devtools/front_end/
H A DnetworkPanel.css140 .network-timing-bar.sending {
/external/chromium_org/third_party/libjingle/source/talk/session/media/
H A Dchannel_unittest.cc550 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());
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/external/chromium_org/third_party/webrtc/voice_engine/
H A Dchannel.h90 sending(false),
98 bool sending; member in struct:webrtc::voe::ChannelState::State
143 state_.sending = enable;
448 return channel_state_.Get().sending;
H A Dchannel.cc490 // 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)
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/
H A Drtp_rtcp.h259 * 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
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/
H A Dmock_rtp_rtcp.h110 int32_t(const bool sending));
114 int32_t(const bool sending));
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Drtp_rtcp_impl.cc135 // 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
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H A Drtp_rtcp_impl.h106 virtual int32_t SetSendingStatus(const bool sending) OVERRIDE;
111 virtual int32_t SetSendingMediaStatus(const bool sending) OVERRIDE;
H A Drtcp_sender.cc214 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
/external/chromium_org/third_party/libjingle/source/talk/media/base/
H A Dvideoengine_unittest.h580 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
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H A Dfakemediaengine.h51 // 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.
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
H A Dwebrtcvideoengine.h246 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
H A Dwebrtcvideoengine2.h273 bool sending() const { return sending_; } function in class:cricket::WebRtcVideoChannel2
H A Dwebrtcvoiceengine.h387 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
H A Dwebrtcvideoengine.cc556 // 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
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/external/chromium_org/native_client_sdk/src/doc/
H A Dfaq.rst334 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
/external/iproute2/doc/
H A Dip-tunnels.tex227 will route \verb|3FFE::/16| via \verb|sit0|, sending all the packets
/external/blktrace/doc/
H A Dblktrace.tex562 start sending requests to the driver. This may happen automatically

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