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    Searched refs:LS_INFO (Results 1 - 25 of 142) sorted by null

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  /external/webrtc/talk/media/base/
cpuid_unittest.cc 37 LOG(LS_INFO) << "ARM: "
39 LOG(LS_INFO) << "NEON: "
41 LOG(LS_INFO) << "X86: "
43 LOG(LS_INFO) << "SSE2: "
45 LOG(LS_INFO) << "SSSE3: "
47 LOG(LS_INFO) << "SSE41: "
49 LOG(LS_INFO) << "SSE42: "
51 LOG(LS_INFO) << "AVX: "
61 LOG(LS_INFO) << "IsCoreIOrBetter: " << core_i_or_better;
  /external/webrtc/webrtc/p2p/stunprober/
main.cc 72 LOG(LS_INFO) << "Shared Socket Mode: " << stats.shared_socket_mode;
73 LOG(LS_INFO) << "Requests sent: " << stats.num_request_sent;
74 LOG(LS_INFO) << "Responses received: " << stats.num_response_received;
75 LOG(LS_INFO) << "Target interval (ns): " << stats.target_request_interval_ns;
76 LOG(LS_INFO) << "Actual interval (ns): " << stats.actual_request_interval_ns;
77 LOG(LS_INFO) << "NAT Type: " << PrintNatType(stats.nat_type);
78 LOG(LS_INFO) << "Host IP: " << stats.host_ip;
79 LOG(LS_INFO) << "Server-reflexive ips: ";
81 LOG(LS_INFO) << "\t" << ip;
84 LOG(LS_INFO) << "Success Precent: " << stats.success_percent
    [all...]
  /external/webrtc/webrtc/base/
linux_unittest.cc 27 LOG(LS_INFO) << "GetNumCpus: " << out_cpus;
32 LOG(LS_INFO) << "GetNumPhysicalCpus: " << out_cpus_phys;
38 LOG(LS_INFO) << "cpu family: " << out_family;
44 LOG(LS_INFO) << "Processor: " << out_processor;
52 LOG(LS_INFO) << "model: " << out_model;
56 LOG(LS_INFO) << "stepping: " << out_stepping;
60 LOG(LS_INFO) << "processor: " << out_processor;
65 LOG(LS_INFO) << "vendor_id: " << out_str;
systeminfo_unittest.cc 18 LOG(LS_INFO) << "CpuVendor: " << info.GetCpuVendor();
40 LOG(LS_INFO) << "CpuArchitecture: " << info.GetCpuArchitecture();
60 LOG(LS_INFO) << "MachineModel: " << machine_model;
84 LOG(LS_INFO) << "MemorySize: " << info.GetMemorySize();
91 LOG(LS_INFO) << "MaxCpus: " << info.GetMaxCpus();
98 LOG(LS_INFO) << "CurCpus: " << info.GetCurCpus();
111 LOG(LS_INFO) << "CpuFamily: " << info.GetCpuFamily();
118 LOG(LS_INFO) << "CpuModel: " << info.GetCpuModel();
125 LOG(LS_INFO) << "CpuStepping: " << info.GetCpuStepping();
133 LOG(LS_INFO) << "CpuFamily: " << info.GetCpuFamily()
    [all...]
logging_unittest.cc 44 LogMessage::AddLogToStream(&stream, LS_INFO);
45 EXPECT_EQ(LS_INFO, LogMessage::GetLogToStream(&stream));
47 LOG(LS_INFO) << "INFO";
66 LogMessage::AddLogToStream(&stream1, LS_INFO);
68 EXPECT_EQ(LS_INFO, LogMessage::GetLogToStream(&stream1));
71 LOG(LS_INFO) << "INFO";
112 LogMessage::AddLogToStream(&stream1, LS_INFO);
152 LOG(LS_INFO) << "Average log time: " << TimeDiff(finish, start) << " us";
macutils_unittest.cc 16 LOG(LS_INFO) << "GetOsVersionName " << ver;
23 LOG(LS_INFO) << "GetQuickTimeVersion " << version;
httprequest.cc 92 LOG(LS_INFO) << "HttpRequest start: " << host_ + client_.request().path;
99 LOG(LS_INFO) << "HttpRequest request timed out";
106 LOG(LS_INFO) << "HttpRequest request error: " << error_;
windowpicker_unittest.cc 25 LOG(LS_INFO) << "skipping test: window capturing is not supported with "
40 LOG(LS_INFO) << "skipping test: window capturing is not supported with "
openssladapter.cc 324 LOG(LS_INFO) << "BeginSSL: " << ssl_host_name_;
398 LOG(LS_INFO) << " -- onStreamReadable";
400 LOG(LS_INFO) << " -- onStreamWriteable";
439 LOG(LS_INFO) << "Cleanup";
466 //LOG(LS_INFO) << "OpenSSLAdapter::Send(" << cb << ")";
494 //LOG(LS_INFO) << " -- success";
497 //LOG(LS_INFO) << " -- error want read";
502 //LOG(LS_INFO) << " -- error want write";
506 //LOG(LS_INFO) << " -- remote side closed";
511 //LOG(LS_INFO) << " -- error " << code
    [all...]
ssladapter_unittest.cc 74 LOG(LS_INFO) << "Initiating connection with " << address;
79 LOG(LS_INFO) << "Starting " << GetSSLProtocolName(ssl_mode_)
95 LOG(LS_INFO) << "Client sending '" << message << "'";
108 LOG(LS_INFO) << "Client received '" << buffer << "'";
148 LOG(LS_INFO) << ((ssl_mode_ == rtc::SSL_MODE_DTLS) ? "UDP" : "TCP")
173 LOG(LS_INFO) << "Server sending '" << message << "'";
223 LOG(LS_INFO) << "Server received '" << buffer << "'";
308 LOG(LS_INFO) << GetSSLProtocolName(ssl_mode_) << " handshake complete.";
315 LOG(LS_INFO) << GetSSLProtocolName(ssl_mode_) << " handshake failed.";
334 LOG(LS_INFO) << "Transfer complete."
    [all...]
  /external/webrtc/webrtc/test/
test_main.cc 25 // Default to LS_INFO, even for release builds to provide better test logging.
27 if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO)
28 rtc::LogMessage::LogToDebug(rtc::LS_INFO);
  /external/webrtc/webrtc/modules/audio_device/ios/
audio_device_ios.mm 40 #define LOGI() LOG(LS_INFO) << "AudioDeviceIOS::"
101 LOG(LS_INFO) << "VerifyAudioSession";
126 LOG(LS_INFO) << "ActivateAudioSession(" << activate << ")";
231 LOG(LS_INFO) << "The audio session is now activated";
234 LOG(LS_INFO) << "Number of audio session users: " << g_audio_session_users;
250 LOG(LS_INFO) << "Our audio session is now deactivated";
253 LOG(LS_INFO) << "Number of audio session users: " << g_audio_session_users;
264 LOG(LS_INFO) << "LogABSD";
265 LOG(LS_INFO) << " sample rate: " << absd.mSampleRate;
266 LOG(LS_INFO) << " format ID: " << formatIDString
    [all...]
  /external/webrtc/talk/app/webrtc/
remotevideocapturer.cc 47 LOG(LS_INFO) << "RemoteVideoCapturer::Start";
59 LOG(LS_INFO) << "RemoteVideoCapturer::Stop";
  /external/webrtc/webrtc/modules/pacing/
bitrate_prober.cc 43 LOG(LS_INFO) << "Initial bandwidth probing enabled";
47 LOG(LS_INFO) << "Initial bandwidth probing disabled";
76 LOG(LS_INFO) << bitrate_log.str().c_str();
109 LOG(LS_INFO) << "Next delta too small, stop probing.";
  /external/webrtc/talk/app/webrtc/test/
peerconnectiontestwrapper.cc 113 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
142 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
155 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
162 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
178 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
190 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
216 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
229 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
240 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
  /external/webrtc/webrtc/modules/audio_processing/agc/
agc_manager_direct.cc 215 LOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
263 LOG(LS_INFO) << "[agc] VolumeCallbacks returned level=0, taking no action.";
273 LOG(LS_INFO) << "[agc] Mic volume was manually adjusted. Updating "
293 LOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", "
307 LOG(LS_INFO) << "[agc] max_level_=" << max_level_
337 LOG(LS_INFO) << "[agc] VolumeCallbacks returned level=0, taking no action.";
344 LOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
349 LOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
400 LOG(LS_INFO) << "[agc] rms_error=" << rms_error << ", "
  /external/webrtc/talk/app/webrtc/java/jni/
androidvideocapturer_jni.cc 65 LOG(LS_INFO) << "AndroidVideoCapturerJni ctor";
70 LOG(LS_INFO) << "AndroidVideoCapturerJni dtor";
79 LOG(LS_INFO) << "AndroidVideoCapturerJni start";
107 LOG(LS_INFO) << "AndroidVideoCapturerJni stop";
119 LOG(LS_INFO) << "AndroidVideoCapturerJni stop done";
146 LOG(LS_INFO) << "AndroidVideoCapturerJni capture started: " << success;
221 LOG(LS_INFO) << "NativeObserver_nativeCapturerStarted";
229 LOG(LS_INFO) << "NativeObserver_nativeOnOutputFormatRequest";
  /external/webrtc/talk/app/webrtc/objc/
RTCLogging.mm 37 return rtc::LS_INFO;
  /external/webrtc/webrtc/base/objc/
RTCLogging.mm 20 return rtc::LS_INFO;
  /external/webrtc/webrtc/system_wrappers/include/
logging.h 15 // NOTE: LS_INFO maps to a new trace level which should be reserved for
18 // impact of adding a new LS_INFO log. If it will be logged at anything
54 // LS_INFO: Chatty level used in debugging for all sorts of things, the default
59 LS_SENSITIVE, LS_VERBOSE, LS_INFO, LS_WARNING, LS_ERROR
101 sev < webrtc::LS_INFO ? (void) 0 :
  /external/webrtc/talk/media/webrtc/
webrtcvoiceengine.cc 281 LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
308 LOG(LS_INFO) << ToString(codec);
506 LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
509 LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!";
521 LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
536 LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
538 LOG(LS_INFO) << ToString(codec);
549 LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
571 LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString();
608 LOG(LS_INFO) << "Always disable AEC and AGC on iOS. Use built-in instead."
    [all...]
  /external/webrtc/webrtc/sound/
automaticallychosensoundsystem.h 65 LOG(LS_INFO) << "Selected " << wrapped_->GetName() << " sound system";
  /external/webrtc/webrtc/system_wrappers/source/
logging.cc 28 case LS_INFO: return kTraceTerseInfo;
  /external/webrtc/webrtc/p2p/base/
dtlstransportchannel.cc 137 LOG_J(LS_INFO, this) << "Ignoring identical DTLS identity";
149 LOG_J(LS_INFO, this) << "NULL DTLS identity supplied. Not doing DTLS";
209 LOG_J(LS_INFO, this) << "Ignoring identical remote DTLS fingerprint";
216 LOG_J(LS_INFO, this) << "Other side didn't support DTLS.";
287 LOG_J(LS_INFO, this) << "Not using DTLS-SRTP.";
290 LOG_J(LS_INFO, this) << "DTLS setup complete.";
466 LOG_J(LS_INFO, this) << "Dropping packet received before DTLS started.";
534 LOG_J(LS_INFO, this) << "DTLS handshake complete.";
553 LOG_J(LS_INFO, this) << "DTLS channel closed";
556 LOG_J(LS_INFO, this) << "DTLS channel error, code=" << err
    [all...]
  /external/webrtc/talk/media/devices/
linuxdevicemanager.cc 215 LOG(LS_INFO) << "Trying " + meta_file_path;
221 LOG(LS_INFO) << "Trying " << meta_file_path;
226 LOG(LS_INFO) << "Trying " << meta_file_path;
236 LOG(LS_INFO) << "Name for " << device_file_name << " is " << device_name;
242 LOG(LS_INFO) << ("Enumerating V4L2 devices");
262 LOG(LS_INFO) << "V4L2 device metadata found at " << metadata_dir;
281 LOG(LS_INFO) << "Plan B. Scanning all video devices in /dev directory";
285 LOG(LS_INFO) << "Total V4L2 devices found : " << devices->size();

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