| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| sctputils.cc | 61 LOG(LS_WARNING) << "Could not read OPEN message type.";
65 LOG(LS_WARNING) << "Data Channel OPEN message of unexpected type: "
72 LOG(LS_WARNING) << "Could not read OPEN message channel type.";
78 LOG(LS_WARNING) << "Could not read OPEN message reliabilility prioirty.";
83 LOG(LS_WARNING) << "Could not read OPEN message reliabilility param.";
88 LOG(LS_WARNING) << "Could not read OPEN message label length.";
93 LOG(LS_WARNING) << "Could not read OPEN message protocol length.";
97 LOG(LS_WARNING) << "Could not read OPEN message label";
101 LOG(LS_WARNING) << "Could not read OPEN message protocol.";
133 LOG(LS_WARNING) << "Could not read OPEN_ACK message type."
[all...] |
| remotevideocapturer.cc | 42 LOG(LS_WARNING)
54 LOG(LS_WARNING)
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtccommon.h | 54 #define LOG_RTCERR0_EX(func, err) LOG(LS_WARNING) \
56 #define LOG_RTCERR1_EX(func, a1, err) LOG(LS_WARNING) \
58 #define LOG_RTCERR2_EX(func, a1, a2, err) LOG(LS_WARNING) \
61 #define LOG_RTCERR3_EX(func, a1, a2, a3, err) LOG(LS_WARNING) \
64 #define LOG_RTCERR4_EX(func, a1, a2, a3, a4, err) LOG(LS_WARNING) \
67 #define LOG_RTCERR5_EX(func, a1, a2, a3, a4, a5, err) LOG(LS_WARNING) \
70 #define LOG_RTCERR6_EX(func, a1, a2, a3, a4, a5, a6, err) LOG(LS_WARNING) \
|
| /external/chromium_org/third_party/libjingle/source/talk/media/devices/ |
| libudevsymboltable.cc | 61 LOG(LS_WARNING)
|
| /external/chromium_org/third_party/webrtc/base/ |
| posix.cc | 113 LOG(LS_WARNING) << "Child reported probles calling chdir()";
117 LOG(LS_WARNING) << "Child reported problems calling fdwalk()";
120 LOG(LS_WARNING) << "Child reported problems calling close()";
|
| atomicops.h | 77 LOG(LS_WARNING) << "Queue capacity is 0.";
97 LOG(LS_WARNING) << "Queue capacity is 0.";
|
| dbus_unittest.cc | 76 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test.";
97 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test.";
125 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test.";
147 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test.";
166 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test.";
191 LOG(LS_WARNING) << "DBus Monitor not started. Skipping test.";
213 LOG(LS_WARNING) << "DBus Monitor not started.";
|
| optionsfile.cc | 30 LOG_F(LS_WARNING) << "Could not open file, err=" << err;
47 LOG_F(LS_WARNING) << "Ignoring malformed line in " << path_;
107 LOG(LS_WARNING) << "Ignoring operation for illegal option " << name;
118 LOG(LS_WARNING) << "Ignoring operation for illegal value " << value;
|
| sslstreamadapterhelper.cc | 83 LOG(LS_WARNING) << "Unknown digest algorithm: " << digest_alg;
96 LOG(LS_WARNING) << "SSLStreamAdapterHelper::Error("
|
| gunit.h | 80 LOG(LS_WARNING) << "Expression " << #ex << " still not true after " << \
|
| /external/chromium_org/jingle/glue/ |
| logging_unittest.cc | 32 case rtc::LS_WARNING:
33 return "LS_WARNING";
79 LOG_V(rtc::LS_WARNING) << AsString(rtc::LS_WARNING);
92 AsString(rtc::LS_WARNING)));
106 LOG_V(rtc::LS_WARNING) << AsString(rtc::LS_WARNING);
119 AsString(rtc::LS_WARNING)));
137 LOG_V(rtc::LS_WARNING) << AsString(rtc::LS_WARNING);
[all...] |
| /external/chromium_org/third_party/webrtc/modules/desktop_capture/win/ |
| screen_capturer_win_magnifier.cc | 92 LOG_F(LS_WARNING) << "Failed to make system & display power assertion: "
125 LOG_F(LS_WARNING) << "Switching to the fallback screen capturer.";
211 LOG_F(LS_WARNING) << "Failed to call SetWindowPos: " << GetLastError()
226 LOG_F(LS_WARNING) << "Failed to call MagSetWindowSource: " << GetLastError()
280 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
287 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
299 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
328 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
344 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
359 LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier:
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
| turnport.cc | 329 LOG(LS_WARNING) << "Socket is bound to a different address then the "
345 LOG_J(LS_WARNING, this) << "Connection with server failed, error=" << error;
353 LOG_J(LS_WARNING, this) << "Giving up on the port after "
465 LOG_J(LS_WARNING, this) << "Received TURN message that was too short";
482 LOG_J(LS_WARNING, this) << "Received TURN message with invalid "
502 LOG_J(LS_WARNING, this) << "Redirection to ["
510 LOG(LS_WARNING) << "Server IP address family does not match with "
555 LOG_J(LS_WARNING, this) << "TURN host lookup received error "
632 LOG_J(LS_WARNING, this) << "Received invalid TURN data indication";
640 LOG_J(LS_WARNING, this) << "Missing STUN_ATTR_XOR_PEER_ADDRESS attribute
[all...] |
| transportdescriptionfactory.cc | 109 LOG(LS_WARNING) << "Failed to create TransportDescription answer "
139 LOG(LS_WARNING) << "Failed to create TransportDescription answer "
|
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtpdataengine.cc | 151 LOG(LS_WARNING) << "Failed to SetRecvCodecs because of unknown codec: "
163 LOG(LS_WARNING) <<
179 LOG(LS_WARNING) << "Not adding data send stream '" << stream.id
216 LOG(LS_WARNING) << "Not adding data recv stream '" << stream.id
238 // LOG(LS_WARNING) << "Could not read rtp header from packet of length "
246 // LOG(LS_WARNING) << "Could not read rtp header"
255 LOG(LS_WARNING) << "Not receiving packet "
265 // LOG(LS_WARNING) << "Not receiving packet "
274 LOG(LS_WARNING) << "Received packet for unknown ssrc: " << header.ssrc;
311 LOG(LS_WARNING) << "Not sending packet with ssrc=" << params.ssr
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtcp_packet.cc | [all...] |
| rtp_utility.cc | 440 LOG(LS_WARNING)
448 LOG(LS_WARNING) << "Failed to find extension id: "
454 LOG(LS_WARNING) << "Incorrect transmission time offset len: "
478 LOG(LS_WARNING) << "Incorrect audio level len: " << len;
500 LOG(LS_WARNING) << "Incorrect absolute send time len: " << len;
517 LOG(LS_WARNING) << "Extension type not implemented: " << type;
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/ |
| Logging.java | 64 LS_SENSITIVE, LS_VERBOSE, LS_INFO, LS_WARNING, LS_ERROR,
|
| /external/chromium_org/third_party/webrtc/system_wrappers/source/ |
| logging.cc | 29 case LS_WARNING: return kTraceWarning;
|
| /external/chromium_org/third_party/webrtc/modules/utility/source/ |
| file_player_impl.cc | 115 LOG(LS_WARNING) << "Get10msAudioFromFile() playing not started!"
175 LOG(LS_WARNING) << "Get10msAudioFromFile() unexpected codec.";
213 LOG(LS_WARNING) << "SetAudioScaling() non-allowed scale factor.";
263 LOG(LS_WARNING) << "StartPlayingFile() failed to initialize "
273 LOG(LS_WARNING) << "StartPlayingFile() failed to initialize "
285 LOG(LS_WARNING) << "StartPlayingFile() failed to initialize file "
401 LOG(LS_WARNING) << "Failed to retrieve codec info of file data.";
407 LOG(LS_WARNING) << "SetUpAudioDecoder() codec " << _codec.plname
589 LOG(LS_WARNING) << "Error reading video data.";
642 LOG(LS_WARNING) << "SetVideoDecoder() failed to retrieve codec info of
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| rtcpmuxfilter.cc | 78 LOG(LS_WARNING) << "Invalid parameters in RTCP mux provisional answer";
95 LOG(LS_WARNING) << "Invalid parameters in RTCP mux answer";
|
| srtpfilter.cc | 212 LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
221 LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
230 LOG(LS_WARNING) << "Failed to ProtectRtcp: SRTP not active";
242 LOG(LS_WARNING) << "Failed to UnprotectRtp: SRTP not active";
250 LOG(LS_WARNING) << "Failed to UnprotectRtcp: SRTP not active";
262 LOG(LS_WARNING) << "Failed to GetRtpAuthParams: SRTP not active";
391 LOG(LS_WARNING) << "Invalid parameters in SRTP answer";
429 LOG(LS_WARNING) << "Failed to apply negotiated SRTP parameters";
496 LOG(LS_WARNING) << "Failed to protect SRTP packet: no SRTP Session";
502 LOG(LS_WARNING) << "Failed to protect SRTP packet: The buffer length
[all...] |
| bundlefilter.cc | 88 LOG(LS_WARNING) << "Stream already added to filter";
|
| channelmanager.cc | 170 LOG(LS_WARNING) << "Cannot toggle rtx after initialization!";
239 LOG(LS_WARNING) << "The preferred microphone '" << audio_in_device_
244 LOG(LS_WARNING) << "The preferred speaker '" << audio_out_device_
250 LOG(LS_WARNING) << "The preferred camera '" << camera_device_
258 LOG(LS_WARNING) << "Failed to SetAudioOptions with"
269 LOG(LS_WARNING) << "Failed to SetOutputVolume to "
273 LOG(LS_WARNING) << "Failed to SetCaptureDevice with camera: "
314 LOG(LS_WARNING) << "failed to delete video capturer";
431 LOG(LS_WARNING) << "Failed to create data channel of type "
440 LOG(LS_WARNING) << "Failed to init data channel."
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/examples/call/ |
| console.cc | 64 LOG(LS_WARNING) << "Already started";
158 LOG(LS_WARNING) << "Can't install signal";
|
