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Searched refs:LS_WARNING (Results 1 – 25 of 162) sorted by relevance

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/external/webrtc/talk/app/webrtc/
Dsctputils.cc58 LOG(LS_WARNING) << "Could not read OPEN message type."; in IsOpenMessage()
73 LOG(LS_WARNING) << "Could not read OPEN message type."; in ParseDataChannelOpenMessage()
77 LOG(LS_WARNING) << "Data Channel OPEN message of unexpected type: " in ParseDataChannelOpenMessage()
84 LOG(LS_WARNING) << "Could not read OPEN message channel type."; in ParseDataChannelOpenMessage()
90 LOG(LS_WARNING) << "Could not read OPEN message reliabilility prioirty."; in ParseDataChannelOpenMessage()
95 LOG(LS_WARNING) << "Could not read OPEN message reliabilility param."; in ParseDataChannelOpenMessage()
100 LOG(LS_WARNING) << "Could not read OPEN message label length."; in ParseDataChannelOpenMessage()
105 LOG(LS_WARNING) << "Could not read OPEN message protocol length."; in ParseDataChannelOpenMessage()
109 LOG(LS_WARNING) << "Could not read OPEN message label"; in ParseDataChannelOpenMessage()
113 LOG(LS_WARNING) << "Could not read OPEN message protocol."; in ParseDataChannelOpenMessage()
[all …]
Dremotevideocapturer.cc42 LOG(LS_WARNING) in Start()
54 LOG(LS_WARNING) in Stop()
/external/webrtc/talk/media/webrtc/
Dwebrtccommon.h53 #define LOG_RTCERR0_EX(func, err) LOG(LS_WARNING) \
55 #define LOG_RTCERR1_EX(func, a1, err) LOG(LS_WARNING) \
57 #define LOG_RTCERR2_EX(func, a1, a2, err) LOG(LS_WARNING) \
60 #define LOG_RTCERR3_EX(func, a1, a2, a3, err) LOG(LS_WARNING) \
63 #define LOG_RTCERR4_EX(func, a1, a2, a3, a4, err) LOG(LS_WARNING) \
66 #define LOG_RTCERR5_EX(func, a1, a2, a3, a4, a5, err) LOG(LS_WARNING) \
69 #define LOG_RTCERR6_EX(func, a1, a2, a3, a4, a5, a6, err) LOG(LS_WARNING) \
Dwebrtcvoiceengine.cc236 LOG(LS_WARNING) << rate_source in GetOpusBitrate()
330 LOG(LS_WARNING) << "Unexpected codec: " << ToString(voe_codec); in SupportedCodecs()
948 sev = rtc::LS_WARNING; in Print()
1000 LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init."; in SetAudioDeviceModule()
1020 LOG(LS_WARNING) << "Could not close file."; in StartAecDump()
1374 LOG(LS_WARNING) << in SetOptions()
1385 LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel"; in SetOptions()
1454 LOG(LS_WARNING) << "Unknown codec " << ToString(codec); in SetRecvCodecs()
1494 LOG(LS_WARNING) << "Unknown codec " << ToString(codec); in SetSendCodecs()
1535 LOG(LS_WARNING) << "Failed to set packet size for codec " in SetSendCodecs()
[all …]
/external/webrtc/webrtc/p2p/base/
Dturnport.cc381 LOG(LS_WARNING) << "Socket is bound to a different address:" in OnSocketConnect()
386 LOG(LS_WARNING) << "Socket is bound to a different address:" in OnSocketConnect()
392 LOG(LS_WARNING) << "Socket is bound to a different address:" in OnSocketConnect()
412 LOG_J(LS_WARNING, this) << "Connection with server failed, error=" << error; in OnSocketClose()
419 LOG_J(LS_WARNING, this) << "Giving up on the port after " in OnAllocateMismatch()
545 LOG_J(LS_WARNING, this) << "Discarding TURN message from unknown address:" in OnReadPacket()
554 LOG_J(LS_WARNING, this) << "Received TURN message that was too short"; in OnReadPacket()
579 LOG_J(LS_WARNING, this) << "Received TURN message with invalid " in OnReadPacket()
604 LOG_J(LS_WARNING, this) << "Redirection to [" in SetAlternateServer()
612 LOG(LS_WARNING) << "Server IP address family does not match with " in SetAlternateServer()
[all …]
/external/webrtc/webrtc/modules/desktop_capture/win/
Dscreen_capturer_win_magnifier.cc93 LOG_F(LS_WARNING) << "Failed to make system & display power assertion: " in Capture()
126 LOG_F(LS_WARNING) << "Switching to the fallback screen capturer."; in Capture()
207 LOG_F(LS_WARNING) << "Failed to call SetWindowPos: " << GetLastError() in CaptureImage()
222 LOG_F(LS_WARNING) << "Failed to call MagSetWindowSource: " << GetLastError() in CaptureImage()
276 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " in InitializeMagnifier()
283 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " in InitializeMagnifier()
295 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " in InitializeMagnifier()
324 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " in InitializeMagnifier()
340 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " in InitializeMagnifier()
355 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: " in InitializeMagnifier()
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/external/webrtc/talk/media/base/
Drtpdataengine.cc136 LOG(LS_WARNING) << "Failed to SetRecvCodecs because of unknown codec: " in SetRecvCodecs()
148 LOG(LS_WARNING) << in SetSendCodecs()
172 LOG(LS_WARNING) << "Not adding data send stream '" << stream.id in AddSendStream()
207 LOG(LS_WARNING) << "Not adding data recv stream '" << stream.id in AddRecvStream()
247 LOG(LS_WARNING) << "Not receiving packet " in OnPacketReceived()
265 LOG(LS_WARNING) << "Received packet for unknown ssrc: " << header.ssrc; in OnPacketReceived()
303 LOG(LS_WARNING) << "Not sending packet with ssrc=" << params.ssrc in SendData()
309 LOG(LS_WARNING) << "Not sending data because binary type is unsupported."; in SendData()
316 LOG(LS_WARNING) << "Not sending data because ssrc is unknown: " in SendData()
323 LOG(LS_WARNING) << "Not sending data because codec is unknown: " in SendData()
/external/webrtc/talk/session/media/
Dsrtpfilter.cc216 LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active"; in ProtectRtp()
229 LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active"; in ProtectRtp()
238 LOG(LS_WARNING) << "Failed to ProtectRtcp: SRTP not active"; in ProtectRtcp()
251 LOG(LS_WARNING) << "Failed to UnprotectRtp: SRTP not active"; in UnprotectRtp()
260 LOG(LS_WARNING) << "Failed to UnprotectRtcp: SRTP not active"; in UnprotectRtcp()
273 LOG(LS_WARNING) << "Failed to GetRtpAuthParams: SRTP not active"; in GetRtpAuthParams()
405 LOG(LS_WARNING) << "Invalid parameters in SRTP answer"; in NegotiateParams()
445 LOG(LS_WARNING) << "Failed to apply negotiated SRTP parameters"; in ApplyParams()
526 LOG(LS_WARNING) << "Failed to protect SRTP packet: no SRTP Session"; in ProtectRtp()
532 LOG(LS_WARNING) << "Failed to protect SRTP packet: The buffer length " in ProtectRtp()
[all …]
/external/webrtc/webrtc/modules/utility/source/
Dfile_player_impl.cc103 LOG(LS_WARNING) << "Get10msAudioFromFile() playing not started!" in Get10msAudioFromFile()
162 LOG(LS_WARNING) << "Get10msAudioFromFile() unexpected codec."; in Get10msAudioFromFile()
200 LOG(LS_WARNING) << "SetAudioScaling() non-allowed scale factor."; in SetAudioScaling()
250 LOG(LS_WARNING) << "StartPlayingFile() failed to initialize " in StartPlayingFile()
260 LOG(LS_WARNING) << "StartPlayingFile() failed to initialize " in StartPlayingFile()
272 LOG(LS_WARNING) << "StartPlayingFile() failed to initialize file " in StartPlayingFile()
388 LOG(LS_WARNING) << "Failed to retrieve codec info of file data."; in SetUpAudioDecoder()
394 LOG(LS_WARNING) << "SetUpAudioDecoder() codec " << _codec.plname in SetUpAudioDecoder()
Dfile_recorder_impl.cc82 LOG(LS_WARNING) << "Failed to initialize file " << fileName in StartRecordingAudioFile()
111 LOG(LS_WARNING) << "Failed to initialize outStream for recording."; in StartRecordingAudioFile()
138 LOG(LS_WARNING) << "RecordAudioToFile() recording audio is not " in RecordAudioToFile()
200 LOG(LS_WARNING) << "RecordAudioToFile() codec " in RecordAudioToFile()
/external/webrtc/webrtc/base/java/src/org/webrtc/
DLogging.java63 LS_SENSITIVE, LS_VERBOSE, LS_INFO, LS_WARNING, LS_ERROR, enumConstant
115 case LS_WARNING: in log()
137 log(Severity.LS_WARNING, tag, message); in w()
147 log(Severity.LS_WARNING, tag, message); in w()
148 log(Severity.LS_WARNING, tag, e.toString()); in w()
149 log(Severity.LS_WARNING, tag, getStackTraceString(e)); in w()
/external/webrtc/webrtc/modules/rtp_rtcp/source/
Drtp_utility.cc322 LOG(LS_WARNING) in ParseOneByteExtensionHeader()
330 LOG(LS_WARNING) << "Failed to find extension id: " << id; in ParseOneByteExtensionHeader()
335 LOG(LS_WARNING) << "Incorrect transmission time offset len: " in ParseOneByteExtensionHeader()
352 LOG(LS_WARNING) << "Incorrect audio level len: " << len; in ParseOneByteExtensionHeader()
368 LOG(LS_WARNING) << "Incorrect absolute send time len: " << len; in ParseOneByteExtensionHeader()
384 LOG(LS_WARNING) in ParseOneByteExtensionHeader()
399 LOG(LS_WARNING) << "Incorrect transport sequence number len: " in ParseOneByteExtensionHeader()
416 LOG(LS_WARNING) << "Extension type not implemented: " << type; in ParseOneByteExtensionHeader()
Dfec_receiver_impl.cc85 LOG(LS_WARNING) << "Corrupt/truncated FEC packet."; in AddReceivedRedPacket()
108 LOG(LS_WARNING) << "Corrupt/truncated FEC packet."; in AddReceivedRedPacket()
118 LOG(LS_WARNING) << "Corrupt payload found."; in AddReceivedRedPacket()
128 LOG(LS_WARNING) << "More than 2 blocks in packet not supported."; in AddReceivedRedPacket()
134 LOG(LS_WARNING) << "Block length longer than packet."; in AddReceivedRedPacket()
Drtp_packet_history.cc44 LOG(LS_WARNING) << "Purging packet history in order to re-set status."; in SetStorePacketsStatus()
90 LOG(LS_WARNING) << "Failed to store RTP packet with length: " in PutRTPPacket()
187 LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number; in GetPacketAndSetSendTime()
194 LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number in GetPacketAndSetSendTime()
/external/webrtc/webrtc/base/
Ddbus_unittest.cc76 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test."; in TEST()
97 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test."; in TEST()
125 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test."; in TEST()
147 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test."; in TEST()
166 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test."; in TEST()
191 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test."; in TEST()
213 LOG(LS_WARNING) << "DBus Monitor not started."; in TEST()
Dposix.cc113 LOG(LS_WARNING) << "Child reported probles calling chdir()"; in RunAsDaemon()
117 LOG(LS_WARNING) << "Child reported problems calling fdwalk()"; in RunAsDaemon()
120 LOG(LS_WARNING) << "Child reported problems calling close()"; in RunAsDaemon()
Doptionsfile.cc32 LOG_F(LS_WARNING) << "Could not open file, err=" << err; in Load()
49 LOG_F(LS_WARNING) << "Ignoring malformed line in " << path_; in Load()
109 LOG(LS_WARNING) << "Ignoring operation for illegal option " << name; in IsLegalName()
120 LOG(LS_WARNING) << "Ignoring operation for illegal value " << value; in IsLegalValue()
/external/webrtc/webrtc/modules/audio_coding/neteq/
Dpayload_splitter.cc92 LOG(LS_WARNING) << "SplitRed length mismatch"; in SplitRed()
135 LOG(LS_WARNING) << "SplitFec unknown payload type"; in SplitFec()
177 LOG(LS_WARNING) << "SplitFec wrong payload type"; in SplitFec()
229 LOG(LS_WARNING) << "SplitAudio unknown payload type"; in SplitAudio()
305 LOG(LS_WARNING) << "SplitAudio too large iLBC payload"; in SplitAudio()
317 LOG(LS_WARNING) << "SplitAudio invalid iLBC payload"; in SplitAudio()
412 LOG(LS_WARNING) << "SplitByFrames length mismatch"; in SplitByFrames()
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/
Dtransport_feedback.cc345 LOG(LS_WARNING) << "Delta value too large ( >= 2^16 ticks )"; in WithReceivedPacket()
385 LOG(LS_WARNING) << "Packet status count too large ( >= 2^16 )"; in Encode()
660 LOG(LS_WARNING) << "Buffer too small (" << length in ParseFrom()
671 LOG(LS_WARNING) << "Invalid RTCP header: FMT must be " in ParseFrom()
678 LOG(LS_WARNING) << "Invalid RTCP header: PT must be " << kPayloadType in ParseFrom()
693 LOG(LS_WARNING) << "Empty feedback messages not allowed."; in ParseFrom()
701 LOG(LS_WARNING) << "Buffer overflow while parsing packet."; in ParseFrom()
723 LOG(LS_WARNING) << "Buffer overflow while parsing packet."; in ParseFrom()
731 LOG(LS_WARNING) << "Buffer overflow while parsing packet."; in ParseFrom()
766 LOG(LS_WARNING) << "Header/body mismatch. " in ParseChunk()
Dbye.cc42 LOG(LS_WARNING) in Parse()
51 LOG(LS_WARNING) << "Invalid reason length: " << reason_length; in Parse()
114 LOG(LS_WARNING) << "Max CSRC size reached."; in WithCsrc()
Dextended_jitter_report.cc50 LOG(LS_WARNING) << "Packet is too small to contain all the jitter."; in Parse()
65 LOG(LS_WARNING) << "Max inter-arrival jitter items reached."; in WithJitter()
Dreceiver_report.cc42 LOG(LS_WARNING) << "Packet is too small to contain all the data."; in Parse()
81 LOG(LS_WARNING) << "Max report blocks reached."; in WithReportBlock()
/external/webrtc/webrtc/modules/audio_device/ios/
Daudio_device_not_implemented_ios.mm37 LOG_F(LS_WARNING) << "Not implemented";
43 LOG_F(LS_WARNING) << "Not implemented";
111 LOG_F(LS_WARNING) << "Not implemented";
180 LOG_F(LS_WARNING) << "Not implemented";
195 LOG_F(LS_WARNING) << "Not implemented";
260 LOG_F(LS_WARNING) << "Not implemented";
/external/webrtc/talk/media/sctp/
Dsctpdataengine.cc511 LOG(LS_WARNING) << debug_name_ << "->Connect(): Ignored as socket " in Connect()
608 LOG(LS_WARNING) << debug_name_ << "->SendData(...): " in SendData()
616 LOG(LS_WARNING) << debug_name_ << "->SendData(...): " in SendData()
720 LOG(LS_WARNING) << debug_name_ << "->OnDataFromSctpToChannel(...): " in OnDataFromSctpToChannel()
733 LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): " in AddStream()
739 LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): " in AddStream()
746 LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): " in AddStream()
837 LOG(LS_WARNING) << "Unknown SCTP event: " in OnNotificationFromSctp()
/external/webrtc/webrtc/modules/video_coding/utility/
Dvp8_header_parser.cc164 LOG(LS_WARNING) << "Failed to get QP, invalid length."; in GetQp()
177 LOG(LS_WARNING) << "Failed to get QP, invalid length: " << length; in GetQp()
195 LOG(LS_WARNING) << "Failed to get QP, end of file reached."; in GetQp()

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