/external/webrtc/webrtc/p2p/stunprober/ |
D | main.cc | 72 LOG(LS_INFO) << "Shared Socket Mode: " << stats.shared_socket_mode; in PrintStats() 73 LOG(LS_INFO) << "Requests sent: " << stats.num_request_sent; in PrintStats() 74 LOG(LS_INFO) << "Responses received: " << stats.num_response_received; in PrintStats() 75 LOG(LS_INFO) << "Target interval (ns): " << stats.target_request_interval_ns; in PrintStats() 76 LOG(LS_INFO) << "Actual interval (ns): " << stats.actual_request_interval_ns; in PrintStats() 77 LOG(LS_INFO) << "NAT Type: " << PrintNatType(stats.nat_type); in PrintStats() 78 LOG(LS_INFO) << "Host IP: " << stats.host_ip; in PrintStats() 79 LOG(LS_INFO) << "Server-reflexive ips: "; in PrintStats() 81 LOG(LS_INFO) << "\t" << ip; in PrintStats() 84 LOG(LS_INFO) << "Success Precent: " << stats.success_percent; in PrintStats() [all …]
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/external/webrtc/talk/media/base/ |
D | cpuid_unittest.cc | 37 LOG(LS_INFO) << "ARM: " in TEST() 39 LOG(LS_INFO) << "NEON: " in TEST() 41 LOG(LS_INFO) << "X86: " in TEST() 43 LOG(LS_INFO) << "SSE2: " in TEST() 45 LOG(LS_INFO) << "SSSE3: " in TEST() 47 LOG(LS_INFO) << "SSE41: " in TEST() 49 LOG(LS_INFO) << "SSE42: " in TEST() 51 LOG(LS_INFO) << "AVX: " in TEST() 61 LOG(LS_INFO) << "IsCoreIOrBetter: " << core_i_or_better; in TEST()
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D | videoadapter.cc | 190 LOG(LS_INFO) << "VAdapt input interval changed from " in SetInputFormat() 208 LOG(LS_INFO) << "VAdapt Input Resolution Change: " in SetInputFormat() 219 LOG(LS_INFO) << "CPU smoothing is now " in set_cpu_smoothing() 232 LOG(LS_INFO) << "VAdapt output interval changed from " in SetOutputFormat() 291 LOG(LS_INFO) << "VAdapt Drop Frame: scaled " << frames_scaled_ in AdaptFrameResolution() 330 LOG(LS_INFO) << "VAdapt Frame: scaled " << frames_scaled_ in AdaptFrameResolution() 353 LOG(LS_INFO) << "Video Adapter third scaling is now " in set_scale_third() 429 LOG(LS_INFO) << "VAdapt View Request: " in OnOutputFormatRequest() 439 LOG(LS_INFO) << "VAdapt Change Cpu Adapt Min Samples from: " in set_cpu_load_min_samples() 451 LOG(LS_INFO) << "VAdapt Change High System Threshold from: " in set_high_system_threshold() [all …]
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/external/webrtc/webrtc/base/ |
D | systeminfo_unittest.cc | 18 LOG(LS_INFO) << "CpuVendor: " << info.GetCpuVendor(); in TEST() 40 LOG(LS_INFO) << "CpuArchitecture: " << info.GetCpuArchitecture(); in TEST() 60 LOG(LS_INFO) << "MachineModel: " << machine_model; in TEST() 84 LOG(LS_INFO) << "MemorySize: " << info.GetMemorySize(); in TEST() 91 LOG(LS_INFO) << "MaxCpus: " << info.GetMaxCpus(); in TEST() 98 LOG(LS_INFO) << "CurCpus: " << info.GetCurCpus(); in TEST() 111 LOG(LS_INFO) << "CpuFamily: " << info.GetCpuFamily(); in TEST() 118 LOG(LS_INFO) << "CpuModel: " << info.GetCpuModel(); in TEST() 125 LOG(LS_INFO) << "CpuStepping: " << info.GetCpuStepping(); in TEST() 133 LOG(LS_INFO) << "CpuFamily: " << info.GetCpuFamily(); in TEST() [all …]
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D | linux_unittest.cc | 27 LOG(LS_INFO) << "GetNumCpus: " << out_cpus; in TEST() 32 LOG(LS_INFO) << "GetNumPhysicalCpus: " << out_cpus_phys; in TEST() 38 LOG(LS_INFO) << "cpu family: " << out_family; in TEST() 44 LOG(LS_INFO) << "Processor: " << out_processor; in TEST() 52 LOG(LS_INFO) << "model: " << out_model; in TEST() 56 LOG(LS_INFO) << "stepping: " << out_stepping; in TEST() 60 LOG(LS_INFO) << "processor: " << out_processor; in TEST() 65 LOG(LS_INFO) << "vendor_id: " << out_str; in TEST()
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D | logging_unittest.cc | 44 LogMessage::AddLogToStream(&stream, LS_INFO); in TEST() 45 EXPECT_EQ(LS_INFO, LogMessage::GetLogToStream(&stream)); in TEST() 47 LOG(LS_INFO) << "INFO"; in TEST() 66 LogMessage::AddLogToStream(&stream1, LS_INFO); in TEST() 68 EXPECT_EQ(LS_INFO, LogMessage::GetLogToStream(&stream1)); in TEST() 71 LOG(LS_INFO) << "INFO"; in TEST() 112 LogMessage::AddLogToStream(&stream1, LS_INFO); in TEST() 152 LOG(LS_INFO) << "Average log time: " << TimeDiff(finish, start) << " us"; in TEST()
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D | openssladapter.cc | 324 LOG(LS_INFO) << "BeginSSL: " << ssl_host_name_; in BeginSSL() 398 LOG(LS_INFO) << " -- onStreamReadable"; in ContinueSSL() 400 LOG(LS_INFO) << " -- onStreamWriteable"; in ContinueSSL() 439 LOG(LS_INFO) << "Cleanup"; in Cleanup() 622 LOG(LS_INFO) << "DTLS timeout expired"; in OnMessage() 630 LOG(LS_INFO) << "OpenSSLAdapter::OnConnectEvent"; in OnConnectEvent() 706 LOG(LS_INFO) << "OpenSSLAdapter::OnCloseEvent(" << err << ")"; in OnCloseEvent() 728 LOG(LS_INFO) << "Certificate from server:"; in VerifyServerName() 734 LOG(LS_INFO) << buffer; in VerifyServerName() 739 LOG(LS_INFO) << "Cipher: " << cipher_description; in VerifyServerName() [all …]
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D | ssladapter_unittest.cc | 74 LOG(LS_INFO) << "Initiating connection with " << address; in Connect() 79 LOG(LS_INFO) << "Starting " << GetSSLProtocolName(ssl_mode_) in Connect() 95 LOG(LS_INFO) << "Client sending '" << message << "'"; in Send() 108 LOG(LS_INFO) << "Client received '" << buffer << "'"; in OnSSLAdapterReadEvent() 148 LOG(LS_INFO) << ((ssl_mode_ == rtc::SSL_MODE_DTLS) ? "UDP" : "TCP") in SSLAdapterTestDummyServer() 173 LOG(LS_INFO) << "Server sending '" << message << "'"; in Send() 223 LOG(LS_INFO) << "Server received '" << buffer << "'"; in OnSSLStreamAdapterEvent() 308 LOG(LS_INFO) << GetSSLProtocolName(ssl_mode_) << " handshake complete."; in TestHandshake() 315 LOG(LS_INFO) << GetSSLProtocolName(ssl_mode_) << " handshake failed."; in TestHandshake() 334 LOG(LS_INFO) << "Transfer complete."; in TestTransfer()
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D | sslstreamadapter_unittest.cc | 69 LOG(LS_INFO) << "Feature disabled... skipping"; \ 116 LOG(LS_INFO) << "SSLDummyStreamBase::OnEvent side=" << side_ << " sig=" in OnEventIn() 125 LOG(LS_INFO) << "SSLDummyStreamBase::OnEvent side=" << side_ << " sig=" in OnEventOut() 142 LOG(LS_INFO) << "Closing outbound stream"; in Close() 317 LOG(LS_INFO) << "SSLStreamAdapterTestBase::OnEvent sig=" << sig; in OnEvent() 333 LOG(LS_INFO) << "Setting peer identities by digest"; in SetPeerIdentitiesByDigest() 340 LOG(LS_INFO) << "Setting bogus digest for server cert"; in SetPeerIdentitiesByDigest() 352 LOG(LS_INFO) << "Setting bogus digest for client cert"; in SetPeerIdentitiesByDigest() 410 LOG(LS_INFO) << "Randomly dropping packet, size=" << data_len; in DataWritten() 415 LOG(LS_INFO) << "Dropping packet > mtu, size=" << data_len; in DataWritten() [all …]
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D | autodetectproxy.cc | 50 LOG(LS_INFO) << "GetProxySettingsForUrl(" << server_url_ << ") - start"; in DoWork() 52 LOG(LS_INFO) << "GetProxySettingsForUrl - stop"; in DoWork() 59 LOG(LS_INFO) << "AutoDetectProxy found proxy at " << proxy_.address; in DoWork() 61 LOG(LS_INFO) << "AutoDetectProxy initiating proxy classification"; in DoWork() 144 LOG(LS_INFO) << "Failed to resolve " << resolver_->address(); in OnResolveResult() 213 LoggingSeverity sev = (proxy_.type == PROXY_UNKNOWN) ? LS_ERROR : LS_INFO; in Complete()
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D | httprequest.cc | 92 LOG(LS_INFO) << "HttpRequest start: " << host_ + client_.request().path; in Send() 99 LOG(LS_INFO) << "HttpRequest request timed out"; in Send() 106 LOG(LS_INFO) << "HttpRequest request error: " << error_; in Send()
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/external/webrtc/talk/media/webrtc/ |
D | webrtcvoiceengine.cc | 281 LOG(LS_INFO) << "WebRtc VoiceEngine codecs:"; in SupportedCodecs() 308 LOG(LS_INFO) << ToString(codec); in SupportedCodecs() 506 LOG(LS_INFO) << "WebRtcVoiceEngine::Init"; in Init() 509 LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!"; in Init() 521 LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString(); in InitInternal() 536 LOG(LS_INFO) << "WebRtc VoiceEngine codecs:"; in InitInternal() 538 LOG(LS_INFO) << ToString(codec); in InitInternal() 549 LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate"; in Terminate() 571 LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString(); in ApplyOptions() 608 LOG(LS_INFO) << "Always disable AEC and AGC on iOS. Use built-in instead."; in ApplyOptions() [all …]
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D | webrtcvideoengine2.cc | 450 LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << "."; in OnUnsignalledSsrc() 477 LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()"; in WebRtcVideoEngine2() 482 LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2"; in ~WebRtcVideoEngine2() 486 LOG(LS_INFO) << "WebRtcVideoEngine2::Init"; in Init() 494 LOG(LS_INFO) << "CreateChannel. Options: " << options.ToString(); in CreateChannel() 715 LOG(LS_INFO) << "SetSendParameters: " << params.ToString(); in SetSendParameters() 736 LOG(LS_INFO) << "SetRecvParameters: " << params.ToString(); in SetRecvParameters() 769 LOG(LS_INFO) << "SetRecvCodecs: " << CodecVectorToString(codecs); in SetRecvCodecs() 790 LOG(LS_INFO) in SetRecvCodecs() 795 LOG(LS_INFO) << "Changing recv codecs from " in SetRecvCodecs() [all …]
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/external/webrtc/webrtc/modules/audio_processing/agc/ |
D | agc_manager_direct.cc | 215 LOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio=" in AnalyzePreProcess() 263 LOG(LS_INFO) << "[agc] VolumeCallbacks returned level=0, taking no action."; in SetLevel() 273 LOG(LS_INFO) << "[agc] Mic volume was manually adjusted. Updating " in SetLevel() 293 LOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", " in SetLevel() 307 LOG(LS_INFO) << "[agc] max_level_=" << max_level_ in SetMaxLevel() 337 LOG(LS_INFO) << "[agc] VolumeCallbacks returned level=0, taking no action."; in CheckVolumeAndReset() 344 LOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level; in CheckVolumeAndReset() 349 LOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level; in CheckVolumeAndReset() 400 LOG(LS_INFO) << "[agc] rms_error=" << rms_error << ", " in UpdateGain()
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/external/webrtc/webrtc/modules/audio_device/ios/ |
D | 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 …]
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/external/webrtc/talk/app/webrtc/test/ |
D | peerconnectiontestwrapper.cc | 113 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ in OnAddStream() 142 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ in OnSuccess() 155 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ in CreateOffer() 162 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ in CreateAnswer() 178 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ in SetLocalDescription() 190 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ in SetRemoteDescription() 216 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ in WaitForConnection() 229 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ in WaitForAudio() 240 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ in WaitForVideo()
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/external/webrtc/talk/app/webrtc/java/jni/ |
D | androidvideocapturer_jni.cc | 65 LOG(LS_INFO) << "AndroidVideoCapturerJni ctor"; in AndroidVideoCapturerJni() 70 LOG(LS_INFO) << "AndroidVideoCapturerJni dtor"; in ~AndroidVideoCapturerJni() 79 LOG(LS_INFO) << "AndroidVideoCapturerJni start"; in Start() 107 LOG(LS_INFO) << "AndroidVideoCapturerJni stop"; in Stop() 119 LOG(LS_INFO) << "AndroidVideoCapturerJni stop done"; in Stop() 146 LOG(LS_INFO) << "AndroidVideoCapturerJni capture started: " << success; in OnCapturerStarted() 221 LOG(LS_INFO) << "NativeObserver_nativeCapturerStarted"; in JOW() 229 LOG(LS_INFO) << "NativeObserver_nativeOnOutputFormatRequest"; in JOW()
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/external/webrtc/talk/media/sctp/ |
D | sctpdataengine.cc | 134 LOG(LS_INFO) << "SCTP: " << s; in debug_sctp_printf() 652 LOG(LS_INFO) << debug_name_ << "->SendData(...): EWOULDBLOCK returned"; in SendData() 699 LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): " in OnInboundPacketFromSctpToChannel() 797 LOG(LS_INFO) << "SCTP_REMOTE_ERROR"; in OnNotificationFromSctp() 800 LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT"; in OnNotificationFromSctp() 803 LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION"; in OnNotificationFromSctp() 806 LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT"; in OnNotificationFromSctp() 809 LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT"; in OnNotificationFromSctp() 817 LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT"; in OnNotificationFromSctp() 820 LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT"; in OnNotificationFromSctp() [all …]
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/external/webrtc/webrtc/p2p/base/ |
D | dtlstransportchannel.cc | 137 LOG_J(LS_INFO, this) << "Ignoring identical DTLS identity"; in SetLocalCertificate() 149 LOG_J(LS_INFO, this) << "NULL DTLS identity supplied. Not doing DTLS"; in SetLocalCertificate() 209 LOG_J(LS_INFO, this) << "Ignoring identical remote DTLS fingerprint"; in SetRemoteFingerprint() 216 LOG_J(LS_INFO, this) << "Other side didn't support DTLS."; in SetRemoteFingerprint() 287 LOG_J(LS_INFO, this) << "Not using DTLS-SRTP."; in SetupDtls() 290 LOG_J(LS_INFO, this) << "DTLS setup complete."; in SetupDtls() 466 LOG_J(LS_INFO, this) << "Dropping packet received before DTLS started."; in OnReadPacket() 534 LOG_J(LS_INFO, this) << "DTLS handshake complete."; in OnDtlsEvent() 553 LOG_J(LS_INFO, this) << "DTLS channel closed"; in OnDtlsEvent() 556 LOG_J(LS_INFO, this) << "DTLS channel error, code=" << err; in OnDtlsEvent() [all …]
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D | pseudotcp.cc | 195 LOG(LS_INFO) << "Stats[" << buffer << "]"; 272 LOG(LS_INFO) << "State: TCP_SYN_SENT"; in Connect() 299 LOG(LS_INFO) << "timeout retransmit (rto: " << m_rx_rto in NotifyClock() 545 LOG(LS_INFO) << "<-- <CONV=" << m_conv in packet() 591 LOG(LS_INFO) << "--> <CONV=" << seg.conv in parse() 686 LOG(LS_INFO) << "State: TCP_SYN_RECEIVED"; in process() 691 LOG(LS_INFO) << "State: TCP_ESTABLISHED"; in process() 730 LOG(LS_INFO) << "rtt: " << rtt in process() 767 LOG(LS_INFO) << "exit recovery"; in process() 772 LOG(LS_INFO) << "recovery retransmit"; in process() [all …]
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D | turnport.cc | 302 LOG_J(LS_INFO, this) << "Trying to connect to TURN server via " in PrepareAddress() 406 LOG(LS_INFO) << "TurnPort connected to " << socket->GetRemoteAddress() in OnSocketConnect() 426 LOG_J(LS_INFO, this) << "Allocating a new socket after " in OnAllocateMismatch() 617 LOG_J(LS_INFO, this) << "Redirecting from TURN server [" in SetAlternateServer() 634 LOG_J(LS_INFO, this) << "Starting TURN host lookup for " in ResolveTurnAddress() 876 LOG_J(LS_INFO, this) << "Scheduled refresh in " << delay << "ms."; in ScheduleRefresh() 1043 LOG_J(LS_INFO, port_) << "TURN allocate request sent" in OnSent() 1049 LOG_J(LS_INFO, port_) << "TURN allocate requested successfully" in OnResponse() 1090 LOG_J(LS_INFO, port_) << "Received TURN allocate error response" in OnErrorResponse() 1178 LOG_J(LS_INFO, port_) << "Applying STUN_ATTR_REALM attribute in " in OnTryAlternate() [all …]
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D | p2ptransportchannel.cc | 375 LOG(LS_INFO) << "Set gather_continually to " << gather_continually_; in SetIceConfig() 381 LOG(LS_INFO) << "Set backup connection ping interval to " in SetIceConfig() 394 LOG(LS_INFO) << "Set ICE receiving timeout to " << receiving_timeout_ in SetIceConfig() 486 LOG(LS_INFO) << "P2PTransportChannel: " << transport_name() << ", component " in OnCandidatesAllocationDone() 599 LOG(LS_INFO) << "Connection already exists for peer reflexive " in OnUnknownAddress() 620 LOG(LS_INFO) << "Adding connection from " in OnUnknownAddress() 659 LOG(LS_INFO) << "Switching best connection on controlled side: " in OnNominated() 667 LOG(LS_INFO) << "Not switching the best connection on controlled side yet," in OnNominated() 798 LOG_J(LS_INFO, this) << "Created connection with origin=" << origin << ", (" in CreateConnection() 1015 LOG(LS_INFO) << "Switching best connection: " << top_connection->ToString(); in SortConnections() [all …]
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D | port.cc | 198 LOG_J(LS_INFO, this) << "Port created"; in Construct() 289 LOG(LS_INFO) << "Received STUN ping " in OnReadPacket() 295 LOG(LS_INFO) << "Received conflicting role from the peer."; in OnReadPacket() 555 LOG_J(LS_INFO, this) in SendBindingResponse() 582 rtc::LS_INFO : rtc::LS_VERBOSE; in SendBindingResponse() 619 LOG_J(LS_INFO, this) << "Sending STUN binding error: reason=" << reason in SendBindingErrorResponse() 659 LOG_J(LS_INFO, this) << "Port deleted"; in Destroy() 795 LOG_J(LS_INFO, this) << "Connection created"; in Connection() 909 rtc::LoggingSeverity sev = (!writable() ? rtc::LS_INFO : rtc::LS_VERBOSE); in OnReadPacket() 965 LOG(LS_INFO) << "Received conflicting role from the peer."; in HandleBindingRequest() [all …]
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/external/webrtc/talk/media/devices/ |
D | linuxdevicemanager.cc | 215 LOG(LS_INFO) << "Trying " + meta_file_path; in GetVideoDeviceName() 221 LOG(LS_INFO) << "Trying " << meta_file_path; in GetVideoDeviceName() 226 LOG(LS_INFO) << "Trying " << meta_file_path; in GetVideoDeviceName() 236 LOG(LS_INFO) << "Name for " << device_file_name << " is " << device_name; in GetVideoDeviceName() 242 LOG(LS_INFO) << ("Enumerating V4L2 devices"); in ScanV4L2Devices() 262 LOG(LS_INFO) << "V4L2 device metadata found at " << metadata_dir; in ScanV4L2Devices() 281 LOG(LS_INFO) << "Plan B. Scanning all video devices in /dev directory"; in ScanV4L2Devices() 285 LOG(LS_INFO) << "Total V4L2 devices found : " << devices->size(); in ScanV4L2Devices()
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/external/webrtc/webrtc/modules/pacing/ |
D | bitrate_prober.cc | 43 LOG(LS_INFO) << "Initial bandwidth probing enabled"; in SetEnabled() 47 LOG(LS_INFO) << "Initial bandwidth probing disabled"; in SetEnabled() 76 LOG(LS_INFO) << bitrate_log.str().c_str(); in MaybeInitializeProbe() 109 LOG(LS_INFO) << "Next delta too small, stop probing."; in TimeUntilNextProbe()
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