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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 "

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