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    Searched refs:LS_ERROR (Results 151 - 175 of 190) sorted by null

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  /external/webrtc/webrtc/p2p/base/
stun.cc 235 LOG(LS_ERROR) << "HMAC computation failed. Message-Integrity "
539 LOG(LS_ERROR) << "Error writing address attribute: unknown family.";
619 LOG(LS_ERROR) << "Error writing xor-address attribute: unknown family.";
797 LOG(LS_ERROR) << "error-code bits not zero";
pseudotcp_unittest.cc 358 LOG_F(LS_ERROR) << "unexpected OnTcpReadable";
568 LOG(LS_ERROR) << "This shouldn't happen - the send buffer is full, "
transport.cc 45 LOG(LS_ERROR) << desc;
p2ptransportchannel.cc 291 LOG(LS_ERROR)
404 LOG(LS_ERROR) << "P2PTransportChannel::Connect: The ice_ufrag_ and the "
    [all...]
relayserver.cc 38 LOG(LS_ERROR) << "SendTo wrote only " << result << " of " << size
41 LOG_ERR(LS_ERROR) << "SendTo";
  /external/webrtc/webrtc/video/
vie_channel.cc 345 LOG_F(LS_ERROR) << "Not a valid send codec " << video_codec.codecType;
349 LOG_F(LS_ERROR) << "Incorrect config "
552 LOG(LS_ERROR) << "Invalid send buffer value.";
732 LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc;
    [all...]
video_send_stream.cc 562 LOG(LS_ERROR) << "Failed to set encoder.";
567 LOG(LS_ERROR) << "Failed to set send codec.";
send_statistics_proxy.cc 255 LOG(LS_ERROR) << "Encoded image outside simulcast range (" << simulcast_idx
  /external/webrtc/talk/session/media/
mediasession.cc     [all...]
planarfunctions_unittest.cc 200 LOG(LS_ERROR) << "Format " << static_cast<int>(fourcc_type)
367 LOG(LS_ERROR) << "Format " << static_cast<int>(fourcc_type)
438 LOG(LS_ERROR) << "Format " << static_cast<int>(fourcc_type)
    [all...]
yuvscaler_unittest.cc 118 LOG(LS_ERROR) << "Failed to load image";
  /external/webrtc/webrtc/modules/video_coding/
codec_database.cc 256 LOG(LS_ERROR) << "Failed to initialize video encoder.";
536 LOG(LS_ERROR) << "Can't find a decoder associated with payload type: "
generic_encoder.cc 127 LOG(LS_ERROR) << "Failed to initialize the encoder associated with "
  /external/webrtc/talk/media/webrtc/
webrtcvideoframe.cc 242 LOG(LS_ERROR) << "Error parsing format: " << GetFourccName(format)
  /external/webrtc/webrtc/base/
base64_unittest.cc 328 LOG(LS_ERROR) << "" # x << " != " # y \
    [all...]
filerotatingstream.cc 118 LOG(LS_ERROR) << "Failed to read from: "
physicalsocketserver.cc 884 LOG_ERR(LS_ERROR) << "pipe failed";
    [all...]
httpcommon.cc 940 LOG(LS_ERROR) << "InitializeSecurityContext returned: "
1006 LOG(LS_ERROR) << "AcquireCredentialsHandle error: "
1017 LOG(LS_ERROR) << "InitializeSecurityContext returned: "
opensslstreamadapter.cc 436 LOG(LS_ERROR) << "Could not find cipher: " << *cipher;
817 LOG(LS_ERROR) << "TLS post connection check failed";
1100 LOG(LS_ERROR) << "SSL_get_verify_result(ssl) = "
    [all...]
win32regkey.cc 1089 LOG_GLE(LS_ERROR) << "AdjustTokenPrivileges failed";
1092 LOG_GLE(LS_ERROR) << "LookupPrivilegeValue failed";
1096 LOG_GLE(LS_ERROR) << "OpenProcessToken(GetCurrentProcess) failed";
  /external/webrtc/webrtc/modules/audio_coding/acm2/
rent_a_codec.cc 176 LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname;
  /external/webrtc/webrtc/modules/audio_processing/
echo_control_mobile_impl.cc 317 LOG(LS_ERROR) << "AECM only supports 16 kHz or lower sample rates";
  /external/webrtc/webrtc/modules/video_coding/codecs/vp9/
vp9_impl.cc 125 LOG(LS_ERROR) << "Multiple temporal layers when manually specifying "
147 LOG(LS_ERROR) << "Scaling factors not specified!";
175 LOG(LS_ERROR) << "Unsupported number of temporal layers: "
    [all...]
  /external/webrtc/webrtc/modules/rtp_rtcp/source/
h264_bitstream_parser.cc 89 LOG_F(LS_ERROR) << "FAILED: " #x; \
  /external/webrtc/webrtc/voice_engine/
transmit_mixer.cc     [all...]

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