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Searched
refs:LS_ERROR
(Results
126 - 150
of
190
) sorted by null
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/external/webrtc/webrtc/modules/desktop_capture/
mouse_cursor_monitor_x11.cc
40
LOG(
LS_ERROR
) << "Failed to query for child windows although window"
window_capturer_mac.mm
202
LOG(
LS_ERROR
) << "Unsupported window image depth: " << bits_per_pixel;
/external/webrtc/webrtc/modules/remote_bitrate_estimator/
overuse_estimator.cc
108
LOG(
LS_ERROR
) << "The over-use estimator's covariance matrix is no longer "
/external/webrtc/webrtc/modules/utility/source/
file_recorder_impl.cc
238
LOG(
LS_ERROR
) << "SetUpAudioEncoder() codec "
helpers_ios.mm
48
LOG(
LS_ERROR
) << StdStringFromNSString(msg);
/external/webrtc/webrtc/video/
video_encoder.cc
66
LOG(
LS_ERROR
) << "Failed to initialize software-encoder fallback.";
/external/webrtc/webrtc/voice_engine/
voice_engine_defines.h
131
LOG_F(
LS_ERROR
) << "not supported"; \
/external/webrtc/talk/media/devices/
win32devicemanager.cc
118
LOG(
LS_ERROR
) << "CoInitialize failed, hr=" << hr;
193
LOG(
LS_ERROR
) << "Failed to create device enumerator, hr=" << hr;
/external/webrtc/webrtc/modules/rtp_rtcp/source/
rtp_receiver_impl.cc
113
LOG(
LS_ERROR
) << "Failed to register payload: " << payload_name << "/"
307
LOG(
LS_ERROR
) << "Failed to create decoder for payload type: "
rtp_format_vp8.cc
672
LOG(
LS_ERROR
) << "Empty payload.";
706
LOG(
LS_ERROR
) << "Error parsing VP8 payload descriptor!";
721
LOG(
LS_ERROR
) << "Error parsing VP8 payload descriptor!";
rtp_receiver_audio.cc
271
LOG(
LS_ERROR
) << "Failed to create decoder for payload type: "
/external/webrtc/webrtc/p2p/base/
turnport.cc
276
LOG(
LS_ERROR
) << "Allocation can't be started without setting the"
292
LOG(
LS_ERROR
) << "IP address family does not match: "
306
LOG(
LS_ERROR
) << "Failed to create TURN client socket";
515
LOG(
LS_ERROR
) << "Did not find the TurnEntry for address " << addr;
678
LOG_J(
LS_ERROR
, this) << "Failed to send TURN message, err="
913
LOG(
LS_ERROR
) << "Missing STUN_ATTR_REALM attribute in "
[
all
...]
pseudotcp.cc
652
LOG_F(
LS_ERROR
) << "wrong conversation";
662
LOG_F(
LS_ERROR
) << "closed";
676
LOG_F(
LS_ERROR
) << "Missing control code";
[
all
...]
/external/webrtc/talk/app/webrtc/
statscollector.cc
780
LOG(
LS_ERROR
) << "Failed to get voice channel stats.";
791
LOG(
LS_ERROR
) << "Failed to get transport name for proxy "
[
all
...]
androidvideocapturer.cc
143
LOG(
LS_ERROR
) << "Failed to parse formats.";
/external/webrtc/talk/media/webrtc/
webrtcvoiceengine.cc
137
LOG(
LS_ERROR
) << "No SSRCs in stream parameters: " << sp.ToString();
141
LOG(
LS_ERROR
) << "Multiple SSRCs in stream parameters: " << sp.ToString();
511
LOG(
LS_ERROR
) << "WebRtcVoiceEngine::Init failed";
[
all
...]
simulcast.cc
167
LOG(
LS_ERROR
) << "SlotSimulcastMaxResolution";
/external/webrtc/webrtc/base/
logging.cc
296
current_level =
LS_ERROR
;
400
case
LS_ERROR
:
autodetectproxy.cc
213
LoggingSeverity sev = (proxy_.type == PROXY_UNKNOWN) ?
LS_ERROR
: LS_INFO;
linux.cc
239
LOG_ERR(
LS_ERROR
) << "Can't call uname()";
natserver.cc
213
LOG(
LS_ERROR
) << "Couldn't find a free port!";
/external/webrtc/webrtc/examples/peerconnection/client/
peer_connection_client.cc
326
LOG(
LS_ERROR
) << "No content length field specified by the server.";
462
LOG(
LS_ERROR
) << "Received error from server";
/external/webrtc/webrtc/modules/audio_coding/acm2/
acm_receiver.cc
178
LOG_F(
LS_ERROR
) << "Payload-type "
394
LOG_F(
LS_ERROR
) << "Cannot remove payload "
/external/webrtc/webrtc/modules/video_capture/
device_info_impl.cc
110
LOG(
LS_ERROR
) << "Invalid deviceCapabilityNumber "
/external/webrtc/webrtc/modules/video_coding/
frame_buffer.cc
116
LOG(
LS_ERROR
) << "Failed to insert packet due to frame being too "
Completed in 452 milliseconds
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