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Searched refs:LS_VERBOSE (Results 1 – 25 of 89) sorted by relevance

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/external/webrtc/modules/audio_device/linux/
Daudio_mixer_manager_alsa_linux.cc46 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in Close()
57 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in CloseSpeaker()
64 RTC_LOG(LS_VERBOSE) << "Closing playout mixer"; in CloseSpeaker()
89 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in CloseMicrophone()
96 RTC_LOG(LS_VERBOSE) << "Closing record mixer"; in CloseMicrophone()
103 RTC_LOG(LS_VERBOSE) << "Closing record mixer 2"; in CloseMicrophone()
110 RTC_LOG(LS_VERBOSE) << "Closing record mixer 3"; in CloseMicrophone()
118 RTC_LOG(LS_VERBOSE) << "Closing record mixer 4"; in CloseMicrophone()
128 RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenSpeaker(name=" in OpenSpeaker()
138 RTC_LOG(LS_VERBOSE) << "Closing playout mixer"; in OpenSpeaker()
[all …]
Daudio_mixer_manager_pulse_linux.cc75 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in SetPulseAudioObjects()
86 RTC_LOG(LS_VERBOSE) << "the PulseAudio objects for the mixer has been set"; in SetPulseAudioObjects()
93 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in Close()
107 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in CloseSpeaker()
118 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in CloseMicrophone()
129 RTC_LOG(LS_VERBOSE) in SetPlayStream()
138 RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetRecStream(recStream)"; in SetRecStream()
146 RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex=" in OpenSpeaker()
160 RTC_LOG(LS_VERBOSE) << "the output mixer device is now open"; in OpenSpeaker()
167 RTC_LOG(LS_VERBOSE) in OpenMicrophone()
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Daudio_device_pulse_linux.cc672 RTC_LOG(LS_VERBOSE) << "number of availiable output devices is " << nDevices; in SetPlayoutDevice()
793 RTC_LOG(LS_VERBOSE) << "number of availiable input devices is " << nDevices; in SetRecordingDevice()
895 RTC_LOG(LS_VERBOSE) << "stream state " in InitPlayout()
1106 RTC_LOG(LS_VERBOSE) << "stopping recording"; in StopRecording()
1126 RTC_LOG(LS_VERBOSE) << "disconnected recording"; in StopRecording()
1222 RTC_LOG(LS_VERBOSE) << "stopping playback"; in StopPlayout()
1242 RTC_LOG(LS_VERBOSE) << "disconnected playback"; in StopPlayout()
1310 RTC_LOG(LS_VERBOSE) << "context state cb"; in PaContextStateCallbackHandler()
1315 RTC_LOG(LS_VERBOSE) << "unconnected"; in PaContextStateCallbackHandler()
1320 RTC_LOG(LS_VERBOSE) << "no state"; in PaContextStateCallbackHandler()
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Daudio_device_alsa_linux.cc603 RTC_LOG(LS_VERBOSE) << "number of available audio output devices is " in SetPlayoutDevice()
672 RTC_LOG(LS_VERBOSE) << "number of availiable audio input devices is " in SetRecordingDevice()
784 RTC_LOG(LS_VERBOSE) << "InitPlayout open (" << deviceName << ")"; in InitPlayout()
839 RTC_LOG(LS_VERBOSE) << "playout snd_pcm_get_params buffer_size:" in InitPlayout()
907 RTC_LOG(LS_VERBOSE) << "InitRecording open (" << deviceName << ")"; in InitRecording()
981 RTC_LOG(LS_VERBOSE) << "capture snd_pcm_get_params, buffer_size:" in InitRecording()
1205 RTC_LOG(LS_VERBOSE) << "handle_playout is now set to NULL"; in StopPlayout()
1303 RTC_LOG(LS_VERBOSE) << "Enum device " << enumCount << " - " << name; in GetDevicesInfo()
1381 RTC_LOG(LS_VERBOSE) << "Trying to recover from " in ErrorRecovery()
1421 RTC_LOG(LS_VERBOSE) << "Recovery - snd_pcm_recover OK"; in ErrorRecovery()
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/external/webrtc/modules/audio_device/win/
Daudio_device_core_win.cc176 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in CoreAudioIsSupported()
216 RTC_LOG(LS_VERBOSE) in CoreAudioIsSupported()
270 RTC_LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::CoreAudioIsSupported()" in CoreAudioIsSupported()
293 RTC_LOG(LS_VERBOSE) << buf; in CoreAudioIsSupported()
296 RTC_LOG(LS_VERBOSE) in CoreAudioIsSupported()
329 RTC_LOG(LS_VERBOSE) << "*** Windows Core Audio is supported ***"; in CoreAudioIsSupported()
331 RTC_LOG(LS_VERBOSE) << "*** Windows Core Audio is NOT supported"; in CoreAudioIsSupported()
407 RTC_LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()" in AudioDeviceWindowsCore()
421 RTC_LOG(LS_VERBOSE) in AudioDeviceWindowsCore()
424 RTC_LOG(LS_VERBOSE) in AudioDeviceWindowsCore()
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/external/webrtc/modules/audio_processing/test/conversational_speech/
Dmock_wavreader_factory.cc48 RTC_LOG(LS_VERBOSE) << "using default parameters for " << filepath; in CreateMock()
55 RTC_LOG(LS_VERBOSE) << "using ad-hoc parameters for " << filepath; in CreateMock()
56 RTC_LOG(LS_VERBOSE) << "sample_rate " << it->second.sample_rate; in CreateMock()
57 RTC_LOG(LS_VERBOSE) << "num_channels " << it->second.num_channels; in CreateMock()
58 RTC_LOG(LS_VERBOSE) << "num_samples " << it->second.num_samples; in CreateMock()
/external/webrtc/rtc_base/
Dopenssl_stream_adapter.cc533 RTC_LOG(LS_VERBOSE) << "OpenSSLStreamAdapter::Write(" << data_len << ")"; in Write()
573 RTC_LOG(LS_VERBOSE) << " -- success"; in Write()
580 RTC_LOG(LS_VERBOSE) << " -- error want read"; in Write()
584 RTC_LOG(LS_VERBOSE) << " -- error want write"; in Write()
602 RTC_LOG(LS_VERBOSE) << "OpenSSLStreamAdapter::Read(" << data_len << ")"; in Read()
640 RTC_LOG(LS_VERBOSE) << " -- success"; in Read()
662 RTC_LOG(LS_VERBOSE) << " -- error want read"; in Read()
665 RTC_LOG(LS_VERBOSE) << " -- error want write"; in Read()
669 RTC_LOG(LS_VERBOSE) << " -- remote side closed"; in Read()
694 RTC_DLOG(LS_VERBOSE) << " -- error " << code; in FlushInput()
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Dlogging_unittest.cc153 RTC_LOG(LS_VERBOSE) << "VERBOSE"; in TEST()
212 LogMessage::AddLogToStream(&stream2, LS_VERBOSE); in TEST()
214 EXPECT_EQ(LS_VERBOSE, LogMessage::GetLogToStream(&stream2)); in TEST()
217 RTC_LOG(LS_VERBOSE) << "VERBOSE"; in TEST()
240 void Run() { RTC_LOG(LS_VERBOSE) << "RTC_LOG"; } in Run()
263 LogMessage::AddLogToStream(&stream3, LS_VERBOSE); in TEST()
320 LogMessage::AddLogToStream(&stream, LS_VERBOSE); in TEST()
323 { LogMessageForTesting sanity_check_msg(__FILE__, __LINE__, LS_VERBOSE); } in TEST()
334 LogMessageForTesting(__FILE__, __LINE__, LS_VERBOSE).stream() << message; in TEST()
Dsocket_adapters.cc234 RTC_LOG(LS_VERBOSE) << "AsyncHttpsProxySocket::Connect(" in Connect()
270 RTC_LOG(LS_VERBOSE) << "AsyncHttpsProxySocket::OnConnectEvent"; in OnConnectEvent()
280 RTC_LOG(LS_VERBOSE) << "AsyncHttpsProxySocket::OnCloseEvent(" << err << ")"; in OnCloseEvent()
354 RTC_LOG(LS_VERBOSE) << "AsyncHttpsProxySocket >> " << str; in SendRequest()
358 RTC_LOG(LS_VERBOSE) << "AsyncHttpsProxySocket << " << data; in ProcessLine()
410 RTC_LOG(LS_VERBOSE) << "Ignoring Proxy-Authenticate: " << auth_method; in ProcessLine()
562 RTC_LOG(LS_VERBOSE) << "Bound on " << addr << ":" << port; in ProcessInput()
569 RTC_LOG(LS_VERBOSE) << "Bound on " << addr << ":" << port; in ProcessInput()
574 RTC_LOG(LS_VERBOSE) << "Bound on <IPV6>:" << port; in ProcessInput()
Dfirewall_socket_server.cc42 RTC_LOG(LS_VERBOSE) << "FirewallSocket outbound TCP connection from " in Connect()
59 RTC_LOG(LS_VERBOSE) << "FirewallSocket outbound packet with type " in SendTo()
82 RTC_LOG(LS_VERBOSE) in RecvFrom()
93 RTC_LOG(LS_VERBOSE) << "FirewallSocket listen attempt denied"; in Listen()
109 RTC_LOG(LS_VERBOSE) << "FirewallSocket inbound TCP connection from " in Accept()
228 RTC_LOG(LS_VERBOSE) << "FirewallSocketServer socket creation denied"; in WrapSocket()
Dssl_stream_adapter_unittest.cc182 RTC_LOG(LS_VERBOSE) << "SSLDummyStreamBase::OnEvent side=" << side_ in OnEventIn()
191 RTC_LOG(LS_VERBOSE) << "SSLDummyStreamBase::OnEvent side=" << side_ in OnEventOut()
386 RTC_LOG(LS_VERBOSE) << "SSLStreamAdapterTestBase::OnEvent sig=" << sig; in OnEvent()
542 RTC_LOG(LS_VERBOSE) << "Randomly dropping packet, size=" << data_len; in DataWritten()
547 RTC_LOG(LS_VERBOSE) << "Dropping packet > mtu, size=" << data_len; in DataWritten()
558 RTC_LOG(LS_VERBOSE) << "Damaging packet"; in DataWritten()
740 RTC_LOG(LS_VERBOSE) << "Sent: " << position + sent; in WriteData()
742 RTC_LOG(LS_VERBOSE) << "Blocked..."; in WriteData()
778 RTC_LOG(LS_VERBOSE) << "Read " << bread; in ReadData()
832 RTC_LOG(LS_VERBOSE) << "Sent: " << sent_; in WriteData()
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Dbase64_unittest.cc362 RTC_LOG(LS_VERBOSE) << "Testing base-64"; in TEST()
374 RTC_LOG(LS_VERBOSE) << "B64: " << base64_tests[i].cyphertext; in TEST()
1353 RTC_LOG(LS_VERBOSE) << "Testing specific base64 file"; in TEST()
Dnetwork_monitor.cc37 RTC_LOG(LS_VERBOSE) << "Network change is received at the network monitor"; in OnNetworksChanged()
/external/webrtc/modules/audio_coding/codecs/opus/
Daudio_encoder_multi_channel_opus_impl.cc196 RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps."; in RecreateEncoderInstance()
199 RTC_LOG(LS_VERBOSE) << "Opus enable FEC"; in RecreateEncoderInstance()
202 RTC_LOG(LS_VERBOSE) << "Opus disable FEC"; in RecreateEncoderInstance()
206 RTC_LOG(LS_VERBOSE) << "Set Opus playback rate to " in RecreateEncoderInstance()
212 RTC_LOG(LS_VERBOSE) << "Set Opus coding complexity to " in RecreateEncoderInstance()
217 RTC_LOG(LS_VERBOSE) << "Opus enable DTX"; in RecreateEncoderInstance()
220 RTC_LOG(LS_VERBOSE) << "Opus disable DTX"; in RecreateEncoderInstance()
225 RTC_LOG(LS_VERBOSE) << "Opus enable CBR"; in RecreateEncoderInstance()
228 RTC_LOG(LS_VERBOSE) << "Opus disable CBR"; in RecreateEncoderInstance()
232 RTC_LOG(LS_VERBOSE) << "Set Opus frame length to " << config_.frame_size_ms in RecreateEncoderInstance()
/external/webrtc/p2p/base/
Dpseudo_tcp_unittest.cc135 RTC_LOG(LS_VERBOSE) << "Opened"; in OnTcpOpen()
149 RTC_LOG(LS_VERBOSE) << "Closed"; in OnTcpClosed()
161 RTC_LOG(LS_VERBOSE) << "Dropping packet due to DropNextPacket, size=" in TcpWritePacket()
167 RTC_LOG(LS_VERBOSE) << "Randomly dropping packet, size=" << len; in TcpWritePacket()
172 RTC_LOG(LS_VERBOSE) << "Dropping packet that exceeds path MTU, size=" in TcpWritePacket()
292 RTC_LOG(LS_VERBOSE) << "Flow Control Lifted"; in OnTcpWriteable()
310 RTC_LOG(LS_VERBOSE) << "Received: " << position; in ReadData()
326 RTC_LOG(LS_VERBOSE) << "Sent: " << position + sent; in WriteData()
329 RTC_LOG(LS_VERBOSE) << "Flow Controlled"; in WriteData()
412 RTC_LOG(LS_VERBOSE) << "Flow Control Lifted"; in OnTcpWriteable()
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Dconnection.cc363 RTC_LOG(LS_VERBOSE) << ToString() << ": set_write_state from: " << old_value in set_write_state()
388 RTC_LOG(LS_VERBOSE) << ToString() << ": set_receiving to " << receiving; in UpdateReceiving()
398 RTC_LOG(LS_VERBOSE) << ToString() << ": set_state"; in set_state()
406 RTC_LOG(LS_VERBOSE) << ToString() << ": Change connected_ to " << value; in set_connected()
482 rtc::LoggingSeverity sev = (!writable() ? rtc::LS_INFO : rtc::LS_VERBOSE); in OnReadPacket()
723 rtc::LoggingSeverity sev = (!writable()) ? rtc::LS_INFO : rtc::LS_VERBOSE; in SendResponseMessage()
754 RTC_LOG(LS_VERBOSE) << ToString() << ": Connection destroyed"; in Destroy()
788 if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE)) { in UpdateState()
791 RTC_LOG(LS_VERBOSE) << ToString() in UpdateState()
853 RTC_LOG(LS_VERBOSE) << ToString() << ": Sending STUN ping, id=" in Ping()
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Dtcp_port.cc172 RTC_LOG(LS_VERBOSE) << "Preparing TCP address, current state: " in PrepareAddress()
287 RTC_LOG(LS_VERBOSE) << ToString() << ": Accepted connection from " in OnNewConnection()
364 RTC_LOG(LS_VERBOSE) << ToString() << ": socket ipaddr: " in TCPConnection()
455 RTC_LOG(LS_VERBOSE) << ToString() << ": Connection established to " in OnConnect()
573 RTC_LOG(LS_VERBOSE) << ToString() << ": Connecting from " in CreateOutgoingTcpSocket()
/external/webrtc/modules/audio_device/mac/
Daudio_mixer_manager_mac.cc62 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in Close()
78 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in CloseSpeakerLocked()
92 RTC_LOG(LS_VERBOSE) << __FUNCTION__; in CloseMicrophoneLocked()
101 RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::OpenSpeaker(id=" << deviceID in OpenSpeaker()
123 RTC_LOG(LS_VERBOSE) << "No process has hogged the output device"; in OpenSpeaker()
127 RTC_LOG(LS_VERBOSE) << "Our process has hogged the output device"; in OpenSpeaker()
153 RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::OpenMicrophone(id=" << deviceID in OpenMicrophone()
171 RTC_LOG(LS_VERBOSE) << "No process has hogged the input device"; in OpenMicrophone()
175 RTC_LOG(LS_VERBOSE) << "Our process has hogged the input device"; in OpenMicrophone()
211 RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::SetSpeakerVolume(volume=" in SetSpeakerVolume()
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Daudio_device_mac.cc90 case rtc::LS_VERBOSE: in logCAMsg()
91 RTC_LOG(LS_VERBOSE) << msg << ": " << err[0] << err[1] << err[2] in logCAMsg()
107 case rtc::LS_VERBOSE: in logCAMsg()
108 RTC_LOG(LS_VERBOSE) << msg << ": " << err[3] << err[2] << err[1] in logCAMsg()
327 RTC_LOG(LS_VERBOSE) << "Hardware model: " << buf; in Init()
811 RTC_LOG(LS_VERBOSE) << "number of available waveform-audio output devices is " in SetPlayoutDevice()
882 RTC_LOG(LS_VERBOSE) << "number of available waveform-audio input devices is " in SetRecordingDevice()
1006 RTC_LOG(LS_VERBOSE) in InitPlayout()
1009 RTC_LOG(LS_VERBOSE) << "MacBook Pro not using internal speakers"; in InitPlayout()
1044 RTC_LOG(LS_VERBOSE) << "Ouput stream format:"; in InitPlayout()
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/external/webrtc/media/sctp/
Dsctp_transport.cc213 if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE) && length > 0) { in VerboseLogPacket()
220 RTC_LOG(LS_VERBOSE) << dump_buf; in VerboseLogPacket()
368 RTC_LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():" in OnSctpOutboundPacket()
575 RTC_LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid in ResetStream()
616 RTC_DLOG(LS_VERBOSE) << "Partially sent message. Buffering the remaining" in SendData()
709 RTC_LOG(LS_VERBOSE) << debug_name_ << "->Connect()."; in Connect()
919 RTC_LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_ in SendQueuedStreamResets()
977 RTC_DLOG(LS_VERBOSE) << "Sending partially buffered message of size " in SendBufferedMessage()
1016 RTC_LOG(LS_VERBOSE) << debug_name_ in OnPacketRead()
1111 RTC_LOG(LS_VERBOSE) << debug_name_ in OnDataOrNotificationFromSctp()
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Dsctp_transport_unittest.cc153 RTC_LOG(LS_VERBOSE) << "Transport setup ----------------------------- "; in SetupConnectedTransportsWithTwoStreams()
157 RTC_LOG(LS_VERBOSE) in SetupConnectedTransportsWithTwoStreams()
526 RTC_LOG(LS_VERBOSE) in TEST_P()
531 RTC_LOG(LS_VERBOSE) << "recv2.received=" << receiver2()->received() in TEST_P()
540 RTC_LOG(LS_VERBOSE) in TEST_P()
546 RTC_LOG(LS_VERBOSE) << "recv1.received=" << receiver1()->received() in TEST_P()
/external/webrtc/video/
Dstream_synchronization.cc70 RTC_LOG(LS_VERBOSE) << "Audio delay: " << current_audio_delay_ms in ComputeDelays()
163 RTC_LOG(LS_VERBOSE) << "Sync video delay " << new_video_delay_ms in ComputeDelays()
/external/webrtc/pc/
Dwebrtc_session_description_factory.cc152 RTC_LOG(LS_VERBOSE) << "DTLS-SRTP disabled."; in WebRtcSessionDescriptionFactory()
160 RTC_LOG(LS_VERBOSE) << "DTLS-SRTP enabled; has certificate parameter."; in WebRtcSessionDescriptionFactory()
179 RTC_LOG(LS_VERBOSE) in WebRtcSessionDescriptionFactory()
481 RTC_LOG(LS_VERBOSE) << "Setting new certificate."; in SetCertificate()
/external/webrtc/rtc_base/java/src/org/webrtc/
DLogging.java91 public enum Severity { LS_VERBOSE, LS_INFO, LS_WARNING, LS_ERROR, LS_NONE } enumConstant
183 log(Severity.LS_VERBOSE, tag, message); in v()
/external/webrtc/sdk/objc/api/logging/
DRTCCallbackLogger.mm51 case rtc::LS_VERBOSE:
124 return rtc::LS_VERBOSE;

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