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    Searched refs:LS_ERROR (Results 126 - 150 of 176) sorted by null

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  /external/chromium_org/third_party/libjingle/source/talk/base/
win32socketserver.cc 232 LOG(LS_ERROR) << "EventSink hwnd is being destroyed, but the event sink"
356 LOG_F(LS_ERROR) << "WSAAsyncGetHostByName error: " << WSAGetLastError();
738 LOG_GLE(LS_ERROR) << "Failed to create message window.";
789 LOG_GLE(LS_ERROR) << "GetMessage failed.";
opensslstreamadapter.cc 298 LOG(LS_ERROR) << "Could not find cipher: " << *cipher;
668 LOG(LS_ERROR) << "TLS post connection check failed";
916 LOG(LS_ERROR) << "SSL_get_verify_result(ssl) = "
autodetectproxy.cc 224 LoggingSeverity sev = (proxy_.type == PROXY_UNKNOWN) ? LS_ERROR : LS_INFO;
linux.cc 257 LOG_ERR(LS_ERROR) << "Can't call uname()";
natserver.cc 166 LOG(LS_ERROR) << "Couldn't find a free port!";
  /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
webrtctexturevideoframe.cc 35 LOG(LS_ERROR) << "Call to unimplemented function "<< __FUNCTION__; \
webrtcvoiceengine.cc 181 case talk_base::LS_ERROR:
310 LOG(LS_ERROR) << "Unable to start soundclip";
510 LOG(LS_ERROR) << "WebRtcVoiceEngine::Init failed";
644 LOG(LS_ERROR) << "Unable to create soundclip: soundclip engine failed to "
    [all...]
webrtcvideoengine.cc 138 case talk_base::LS_ERROR:
207 LOG(LS_ERROR)
874 LOG(LS_ERROR) << "Failed to initialize list of supported codec types";
907 LOG(LS_ERROR) << "Failed to start CPU monitor.";
915 LOG(LS_ERROR) << "VideoEngine Init failed, releasing";
    [all...]
  /external/chromium/third_party/libjingle/source/talk/base/
socketaddress.cc 218 LOG_F(LS_ERROR) << "(" << hostname_ << ") err: " << errcode;
354 LOG(LS_ERROR) << "gethostbyname err: " << errcode;
opensslstreamadapter.cc 469 LOG(LS_ERROR) << "TLS post connection check failed";
639 LOG(LS_ERROR) << "SSL_get_verify_result(ssl) = "
physicalsocketserver.cc 645 LOG_ERR(LS_ERROR) << "pipe failed";
    [all...]
win32socketserver.cc 346 LOG_F(LS_ERROR) << "WSAAsyncGetHostByName error: " << WSAGetLastError();
699 LOG_GLE(LS_ERROR) << "Failed to create message window.";
  /external/chromium_org/third_party/libjingle/source/talk/media/devices/
win32devicemanager.cc 107 LOG(LS_ERROR) << "CoInitialize failed, hr=" << hr;
182 LOG(LS_ERROR) << "Failed to create device enumerator, hr=" << hr;
devicemanager.cc 205 LOG_F(LS_ERROR) << " should never be called!";
  /external/chromium_org/third_party/libjingle/source/talk/p2p/client/
httpportallocator.cc 162 LOG(LS_ERROR) << "HttpPortAllocator: maximum number of requests reached; "
168 LOG(LS_ERROR) << "HttpPortAllocator: no relay hosts configured.";
  /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/
statscollector.cc 714 LOG(LS_ERROR) << "Failed to get voice channel stats.";
720 LOG(LS_ERROR) << "Failed to get transport name for proxy "
734 LOG(LS_ERROR) << "Failed to get video channel stats.";
740 LOG(LS_ERROR) << "Failed to get transport name for proxy "
747 LOG(LS_ERROR) << "BWEs count: " << video_info.bw_estimations.size();
767 LOG(LS_ERROR) << "No transport ID mapping for " << proxy;
    [all...]
  /external/chromium_org/third_party/libjingle/source/talk/p2p/base/
turnport.cc 203 LOG(LS_ERROR) << "Allocation can't be started without setting the"
219 LOG(LS_ERROR) << "Server IP address family does not match with "
424 LOG_J(LS_ERROR, this) << "Failed to send TURN message, err="
596 LOG(LS_ERROR) << "Missing STUN_ATTR_REALM attribute in "
605 LOG(LS_ERROR) << "Missing STUN_ATTR_NONCE attribute in "
    [all...]
p2ptransport.cc 121 LOG(LS_ERROR) << "Failed to serialize non-GICE TransportDescription";
pseudotcp.cc 670 LOG_F(LS_ERROR) << "wrong conversation";
680 LOG_F(LS_ERROR) << "closed";
694 LOG_F(LS_ERROR) << "Missing control code";
    [all...]
stun.cc 251 LOG(LS_ERROR) << "HMAC computation failed. Message-Integrity "
553 LOG(LS_ERROR) << "Error writing address attribute: unknown family.";
632 LOG(LS_ERROR) << "Error writing xor-address attribute: unknown family.";
809 LOG(LS_ERROR) << "error-code bits not zero";
pseudotcp_unittest.cc 376 LOG_F(LS_ERROR) << "unexpected OnTcpReadable";
586 LOG(LS_ERROR) << "This shouldn't happen - the send buffer is full, "
  /external/chromium_org/third_party/libjingle/source/talk/session/media/
mediasession.cc     [all...]
call.cc 185 LOG(LS_ERROR) << "Couldn't write out view request: " << error.text;
345 LOG(LS_ERROR) << "Unable to stop screencast with ssrc " << ssrc;
874 LOG(LS_ERROR) << "Failure in audio SetRemoteContent with CA_UPDATE";
885 LOG(LS_ERROR) << "Failure in video SetRemoteContent with CA_UPDATE";
896 LOG(LS_ERROR) << "Failure in data SetRemoteContent with CA_UPDATE";
    [all...]
  /external/chromium/third_party/libjingle/source/talk/session/tunnel/
pseudotcpchannel.cc 504 LOG_F(LS_ERROR) << "EMSGSIZE";
507 PLOG(LS_ERROR, channel_->GetError()) << "PseudoTcpChannel::TcpWritePacket";
  /external/chromium_org/third_party/libjingle/source/talk/examples/peerconnection/client/
peer_connection_client.cc 343 LOG(LS_ERROR) << "No content length field specified by the server.";
479 LOG(LS_ERROR) << "Received error from server";

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