| /external/webrtc/webrtc/p2p/base/ |
| tcpport.cc | 183 LOG_J(LS_INFO, this) << "Not listening due to firewall restrictions.";
416 LOG_J(LS_INFO, this) << "Connection closed with error " << error;
463 LOG_J(LS_INFO, this) << "TCP Connection with remote is closed, "
|
| turnserver.cc | 457 LOG(LS_INFO) << "Sending error response, type=" << resp.type()
579 LOG_J(LS_INFO, this) << "Allocation destroyed";
630 LOG_J(LS_INFO, this) << "Created allocation, lifetime=" << lifetime_secs;
658 LOG_J(LS_INFO, this) << "Refreshed allocation, lifetime=" << lifetime_secs;
710 LOG_J(LS_INFO, this) << "Created permission, peer="
759 LOG_J(LS_INFO, this) << "Bound channel, id=" << channel_id
|
| relayserver.cc | 222 LOG(LS_INFO) << "Dropping packet: no external connection";
369 LOG(LS_INFO) << "Added new binding " << username << ", "
526 LOG(LS_INFO) << "Removed binding " << binding->username() << ", "
738 LOG(LS_INFO) << "Expiring binding " << username_;
|
| stunrequest_unittest.cc | 156 LOG(LS_INFO) << "STUN request #" << (i + 1)
|
| /external/webrtc/talk/app/webrtc/ |
| dtmfsender.cc | 248 LOG(LS_INFO) << "The Dtmf provider is deleted. Clear the sending queue.";
|
| datachannel.cc | 362 LOG(LS_INFO) << "DataChannel received OPEN_ACK message, sid = "
638 LOG(LS_INFO) << "Sent CONTROL message on channel " << config_.id;
|
| dtlsidentitystore.cc | 70 LOG(LS_INFO) << "Generating identity, using keytype " << key_type_;
|
| /external/webrtc/talk/media/devices/ |
| devicemanager_unittest.cc | 386 LOG(LS_INFO) << "skipping test: window capturing is not supported with "
395 LOG(LS_INFO) << "skipping test: window capturing. Does not have any "
410 LOG(LS_INFO) << "skipping test: desktop capturing is not supported with "
419 LOG(LS_INFO) << "skipping test: desktop capturing. Does not have any "
|
| /external/webrtc/talk/session/media/ |
| currentspeakermonitor.cc | 187 LOG(LS_INFO) << "Current speaker changed to " << current_speaker_ssrc_;
|
| yuvscaler_unittest.cc | 158 LOG(LS_INFO) << "Time: " << std::setw(9) << t;
171 LOG(LS_INFO) << "Image MSE: " <<
|
| /external/webrtc/webrtc/base/ |
| systeminfo.cc | 82 LOG(LS_INFO) << "Available number of cores: " << number_of_cores;
|
| network_unittest.cc | 767 LOG(LS_INFO) << "Looking for dummy network: ";
770 LOG(LS_INFO) << " Network name: " << (*it)->name();
778 LOG(LS_INFO) << "No dummy found, quitting.";
781 LOG(LS_INFO) << "Found dummy, running again while ignoring non-default "
787 LOG(LS_INFO) << " Network name: " << (*it)->name();
[all...] |
| fileutils.cc | 238 LOG(LS_INFO) << "Path " << path.pathname() << std::endl;
|
| natserver.cc | 191 LOG(LS_INFO) << "Packet from " << remote_addr.ToSensitiveString()
|
| /external/webrtc/webrtc/modules/desktop_capture/ |
| mouse_cursor_monitor_x11.cc | 126 LOG(LS_INFO) << "X server does not support XFixes.";
|
| screen_capturer_mac.mm | 804 LOG(LS_INFO) << "Using CgBlitPostLion.";
838 LOG(LS_INFO) << "Using CgBlitPreLion (Multi-monitor).";
844 LOG(LS_INFO) << "Using CgBlitPreLion (OpenGL unavailable).";
848 LOG(LS_INFO) << "Using GlBlit";
[all...] |
| window_capturer_mac.mm | 145 LOG(LS_INFO) << "Window not found";
|
| /external/webrtc/talk/app/webrtc/objc/ |
| RTCFileLogger.mm | 180 return rtc::LS_INFO;
|
| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| transient_suppressor.cc | 298 LOG(LS_INFO) << "[ts] Transient suppression is now enabled.";
307 LOG(LS_INFO) << "[ts] Transient suppression is now disabled.";
|
| /external/webrtc/webrtc/p2p/client/ |
| basicportallocator.cc | 438 LOG(LS_INFO) << "Adding allocated port for " << content_name();
469 LOG_J(LS_INFO, port) << "Added port to allocator";
649 LOG(LS_INFO) << "All candidates gathered for " << content_name_ << ":"
661 LOG_J(LS_INFO, port) << "Removed port from allocator ("
814 LOG_J(LS_INFO, network_) << "Allocation Phase="
905 LOG(LS_INFO) << "AllocationSequence: UDPPort will be handling the "
[all...] |
| httpportallocator.cc | 160 LOG(LS_INFO) << "HTTPPortAllocator: sending to relay host " << host;
|
| /external/webrtc/webrtc/video/ |
| video_send_stream.cc | 132 LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
246 LOG(LS_INFO) << "~VideoSendStream: " << config_.ToString();
300 LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
|
| video_receive_stream.cc | 154 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
290 LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
|
| /external/webrtc/talk/media/webrtc/ |
| simulcast.cc | 173 LOG(LS_INFO) << "SlotSimulcastMaxResolution to width:" << *width
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_single_stream.cc | 59 LOG(LS_INFO) << "RemoteBitrateEstimatorSingleStream: Instantiating.";
|
