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Searched
refs:LS_ERROR
(Results
51 - 75
of
190
) sorted by null
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/external/webrtc/webrtc/modules/desktop_capture/
screen_capturer_x11.cc
149
LOG(
LS_ERROR
) << "Unable to get the root window";
156
LOG(
LS_ERROR
) << "Unable to get graphics context";
176
LOG(
LS_ERROR
) << "Failed to initialize pixel buffer.";
209
LOG(
LS_ERROR
) << "Unable to initialize XDamage.";
217
LOG(
LS_ERROR
) << "Unable to create XFixes region.";
385
LOG(
LS_ERROR
) << "Failed to initialize pixel buffer after screen "
/external/webrtc/talk/app/webrtc/objc/
RTCVideoCapturer.mm
49
LOG(
LS_ERROR
) << "GetVideoCaptureDevice failed";
/external/webrtc/talk/app/webrtc/
webrtcsessiondescriptionfactory.cc
263
LOG(
LS_ERROR
) << error;
270
LOG(
LS_ERROR
) << error;
293
LOG(
LS_ERROR
) << error;
299
LOG(
LS_ERROR
) << error;
306
LOG(
LS_ERROR
) << error;
313
LOG(
LS_ERROR
) << error;
486
LOG(
LS_ERROR
) << "Create SDP failed: " << error;
500
LOG(
LS_ERROR
) << "Async identity request failed: error = " << error;
rtpsender.cc
102
LOG(
LS_ERROR
) << "SetTrack can't be called on a stopped RtpSender.";
106
LOG(
LS_ERROR
) << "SetTrack called on audio RtpSender with " << track->kind()
235
LOG(
LS_ERROR
) << "SetTrack can't be called on a stopped RtpSender.";
239
LOG(
LS_ERROR
) << "SetTrack called on video RtpSender with " << track->kind()
datachannel.cc
164
LOG(
LS_ERROR
) << "Failed to initialize the RTP data channel due to "
173
LOG(
LS_ERROR
) << "Failed to initialize the SCTP data channel due to "
178
LOG(
LS_ERROR
) <<
389
LOG(
LS_ERROR
) << "Queued received data exceeds the max buffer size.";
582
LOG(
LS_ERROR
) << "Closing the DataChannel due to a failure to send data, "
592
LOG(
LS_ERROR
) << "Can't buffer any more data for the data channel.";
648
LOG(
LS_ERROR
) << "Closing the DataChannel due to a failure to send"
webrtcsession.cc
199
LOG(
LS_ERROR
) << kInvalidSdp;
236
LOG(
LS_ERROR
) << kInvalidSdp;
241
LOG(
LS_ERROR
) << "Session description must have ice ufrag and pwd.";
278
LOG(
LS_ERROR
) << "Audio not used in this call";
346
LOG(
LS_ERROR
) << desc.str();
[
all
...]
/external/webrtc/talk/media/devices/
yuvframescapturer.cc
137
LOG(
LS_ERROR
) << "Yuv Frame Generator is already running";
151
LOG(
LS_ERROR
) << "Yuv Frame Generator failed to start";
/external/webrtc/talk/media/webrtc/
webrtcmediaengine.cc
100
LOG(
LS_ERROR
) << "Bad RTP extension ID: " << extension.ToString();
104
LOG(
LS_ERROR
) << "Duplicate RTP extension ID: " << extension.ToString();
/external/webrtc/webrtc/base/
win32window.cc
44
LOG_GLE(
LS_ERROR
) << "GetModuleHandleEx failed";
57
LOG_GLE(
LS_ERROR
) << "RegisterClassEx failed";
diskcache_win32.cc
71
LOG_F(
LS_ERROR
) << "Couldn't delete cache files";
helpers.cc
210
LOG(
LS_ERROR
) << "Failed to init random generator!";
228
LOG(
LS_ERROR
) << "Failed to generate random string!";
255
LOG(
LS_ERROR
) << "Failed to generate random string!";
286
LOG(
LS_ERROR
) << "Failed to generate random id!";
macsocketserver.cc
274
LOG_E(
LS_ERROR
, OS, result) << "SendEventToEventTarget";
304
LOG_E(
LS_ERROR
, OS, result) << "PostEventToQueue";
355
LOG(
LS_ERROR
) << "Failed setting next fire time.";
380
LOG_E(
LS_ERROR
, OS, result) << "PostEventToQueue";
macwindowpicker.cc
53
LOG(
LS_ERROR
) << "Could not load CoreGraphics";
165
LOG(
LS_ERROR
) << "Failed getting process for pid";
170
LOG(
LS_ERROR
) << "Failed setting process to front";
188
LOG_E(
LS_ERROR
, OS, err) << "Failed to enumerate the active displays.";
proxydetect.cc
387
LOG(
LS_ERROR
) << "SHGetFolderPath failed";
397
LOG(
LS_ERROR
) << "FSFindFolder failed";
403
LOG(
LS_ERROR
) << "FSRefMakePath failed";
504
LOG(
LS_ERROR
) << "Failed to open file: " << filename.pathname();
657
LOG(
LS_ERROR
) << "Failed to load winhttp.dll.";
709
LOG(
LS_ERROR
) << "Failed to load winhttp.dll.";
783
LOG(
LS_ERROR
) << "Failed loading WinHTTP functions.";
[
all
...]
/external/webrtc/webrtc/modules/rtp_rtcp/source/
rtp_receiver_video.cc
77
LOG(
LS_ERROR
) << "Failed to create depacketizer.";
121
LOG(
LS_ERROR
) << "Failed to created decoder for payload type: "
/external/webrtc/webrtc/voice_engine/
utility.cc
55
LOG(
LS_ERROR
) << "InitializeIfNeeded failed: sample_rate_hz = "
66
LOG(
LS_ERROR
) << "Resample failed: audio_ptr = " << audio_ptr
/external/webrtc/talk/app/webrtc/java/jni/
androidmediacodeccommon.h
52
#define ALOGE LOG_TAG(rtc::
LS_ERROR
, TAG)
/external/webrtc/talk/media/sctp/
sctpdataengine.cc
230
LOG(
LS_ERROR
) << "Received an unknown PPID " << ppid
268
LOG(
LS_ERROR
) << "Got different send size than expected: " << send_size;
316
LOG(
LS_ERROR
) << "Failed to shutdown usrsctp.";
358
LOG(
LS_ERROR
) << "SendThresholdCallback: Failed to get channel for socket "
414
LOG_ERRNO(
LS_ERROR
) << debug_name_ << "Failed to create SCTP socket.";
421
LOG_ERRNO(
LS_ERROR
) << debug_name_ << "Failed to set SCTP to non blocking.";
433
LOG_ERRNO(
LS_ERROR
) << debug_name_ << "Failed to set SO_LINGER.";
443
LOG_ERRNO(
LS_ERROR
) << debug_name_
452
LOG_ERRNO(
LS_ERROR
) << debug_name_ << "Failed to set SCTP_NODELAY.";
463
LOG_ERRNO(
LS_ERROR
) << debug_name
[
all
...]
/external/webrtc/webrtc/libjingle/xmpp/
pingtask.cc
50
LOG(
LS_ERROR
) << "ping_period_millis should be >= ping_timeout_millis";
/external/webrtc/webrtc/modules/desktop_capture/win/
cursor.cc
115
LOG_F(
LS_ERROR
) << "Unable to get cursor icon info. Error = "
131
LOG_F(
LS_ERROR
) << "Unable to get bitmap info. Error = "
159
LOG_F(
LS_ERROR
) << "Unable to get bitmap bits. Error = "
179
LOG_F(
LS_ERROR
) << "Unable to get bitmap bits. Error = "
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/
report_block.cc
51
LOG(
LS_ERROR
) << "Report Block should be 24 bytes long";
/external/webrtc/webrtc/system_wrappers/include/
logging.h
57
//
LS_ERROR
: Something that should not have occurred.
59
LS_SENSITIVE, LS_VERBOSE, LS_INFO, LS_WARNING,
LS_ERROR
/external/webrtc/webrtc/examples/peerconnection/client/
conductor.cc
190
LOG(
LS_ERROR
) << "Failed to serialize candidate";
243
LOG(
LS_ERROR
) << "Failed to initialize our PeerConnection instance";
269
LOG(
LS_ERROR
) << "Failed to initialize our PeerConnection instance";
375
LOG(
LS_ERROR
) << "Can't create device manager";
380
LOG(
LS_ERROR
) << "Can't enumerate video devices";
414
LOG(
LS_ERROR
) << "Adding stream to PeerConnection failed";
468
LOG(
LS_ERROR
) << "SendToPeer failed";
/external/webrtc/talk/media/base/
capturemanager.cc
241
LOG(
LS_ERROR
) << "RestartVideoCapture: video_capturer is not registered.";
246
LOG(
LS_ERROR
) << "RestartVideoCapture: unable to start video capture with "
252
LOG(
LS_ERROR
) << "RestartVideoCapture: unable to stop video capture with "
270
LOG(
LS_ERROR
) << "RestartVideoCapture: Restart failed.";
284
LOG(
LS_ERROR
) << "Unknown/unimplemented RestartOption";
/external/webrtc/webrtc/p2p/base/
stunport.cc
56
LOG(
LS_ERROR
) << "Binding response missing mapped address.";
59
LOG(
LS_ERROR
) << "Binding address has bad family";
77
LOG(
LS_ERROR
) << "Bad allocate response error code";
79
LOG(
LS_ERROR
) << "Binding error response:"
96
LOG(
LS_ERROR
) << "Binding request timed out from "
282
LOG_J(
LS_ERROR
, this) << "UDP send of " << size
Completed in 831 milliseconds
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