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Functions1,316 in github.com/GrowthEase/LLS-Player

⨍ Functions 1,316◇ Types & classes 335

MethodSetMaximumDelay

src/modules/audio_coding/neteq/decision_logic.h:88

MethodSetMaximumDelay

src/modules/audio_coding/acm2/acm_receiver.cc:73

MethodSetMicrophoneMute

src/modules/audio_device/include/fake_audio_device_impl.h:94

MethodSetMicrophoneVolume

src/modules/audio_device/include/fake_audio_device_impl.h:82

MethodSetMinimumDelay

src/modules/audio_coding/neteq/delay_manager.cc:285

MethodSetMinimumDelay

src/modules/audio_coding/neteq/decision_logic.h:91

MethodSetMinimumPlayoutDelay

src/audio/channel_receive.cc:965

MethodSetMinimumPlayoutDelay

src/audio/audio_receive_stream.cc:425

MethodSetOutputVolume

src/media/engine/webrtc_voice_engine.cc:1274

MethodSetPacketLossRate

src/modules/audio_coding/acm2/audio_coding_module.cc:523

MethodSetPacketSampleRateHz

src/modules/audio_coding/codecs/aac/audio_decoder_aac.h:37

MethodSetParametersForPassiveSpeech

src/modules/audio_coding/neteq/accelerate.cc:38

MethodSetParametersForPassiveSpeech

src/modules/audio_coding/neteq/preemptive_expand.cc:45

MethodSetPayloadTypeDemuxingEnabled

src/pc/channel.cc:296

MethodSetPayloadTypeDemuxingEnabled_w

src/pc/channel.cc:579

MethodSetPlayout

src/media/engine/webrtc_voice_engine.cc:1279

MethodSetPlayoutDevice

Device selection

src/modules/audio_device/include/fake_audio_device_impl.h:46

MethodSetRawAudioSink

src/media/engine/webrtc_voice_engine.cc:1266

MethodSetReadyToSend

src/pc/rtp_transport.cc:291

MethodSetReceiveCodecs

src/modules/audio_coding/acm2/audio_coding_module.cc:553

MethodSetRecordableEncodedFrameCallback

src/media/engine/webrtc_video_engine.cc:3451

MethodSetRecordingDevice

src/modules/audio_device/include/fake_audio_device_impl.h:48

MethodSetRecvCodecs

src/media/engine/webrtc_voice_engine.cc:1588

MethodSetRecvParameters

src/media/engine/webrtc_video_engine.cc:1220

MethodSetRemoteContent

src/pc/channel.cc:288

MethodSetRemoteContent_w

src/pc/channel.cc:883

MethodSetRtcpMuxEnabled

src/pc/rtp_transport.cc:30

MethodSetRtcpPacketTransport

src/pc/rtp_transport.cc:86

MethodSetRtpExtensions

src/media/engine/webrtc_voice_engine.cc:1249

MethodSetRtpExtensions

src/audio/audio_receive_stream.cc:253

MethodSetRtpPacketTransport

src/pc/rtp_transport.cc:50

MethodSetRtpSendParameters

src/media/engine/webrtc_voice_engine.cc:1469

MethodSetRtpSendParameters

src/media/engine/webrtc_video_engine.cc:1075

MethodSetRtpTransport

src/pc/channel.cc:223

MethodSetSampleRateAndChannels

src/modules/audio_coding/neteq/neteq_impl.cc:2241

src/media/engine/webrtc_voice_engine.cc:943

MethodSetSendCodecs

Utility function called from SetSendParameters() to extract current send codec settings from the given list of codecs (originally from SDP). Both send

src/media/engine/webrtc_voice_engine.cc:1762

src/audio/channel_receive.cc:613

src/audio/audio_receive_stream.cc:355

MethodSetSpeakerMute

src/modules/audio_device/include/fake_audio_device_impl.h:89

MethodSetSpeakerVolume

src/modules/audio_device/include/fake_audio_device_impl.h:75

MethodSetStereoPlayout

src/modules/audio_device/include/fake_audio_device_impl.h:99

MethodSetStereoRecording

src/modules/audio_device/include/fake_audio_device_impl.h:102

src/video/video_receive_stream2.cc:345

MethodSetSyncGroup

src/audio/audio_receive_stream.cc:447

MethodSetTimingFrameInfo

src/modules/video_coding/timing.cc:317

MethodSetVideoCodecSwitchingEnabled

src/media/engine/webrtc_video_engine.cc:1946

MethodSetVideoSend

src/media/engine/webrtc_video_engine.cc:1291

MethodSignalNetworkState

src/video/video_receive_stream2.cc:335

MethodSignalNetworkState

src/video/rtp_video_stream_receiver2.cc:1098

MethodSignalSentPacket_n

src/pc/channel.cc:778

src/media/engine/unhandled_packets_buffer.h:47

src/modules/video_coding/frame_buffer2.cc:382

src/modules/audio_coding/neteq/neteq_impl.cc:2559

MethodSpeakerIsInitialized

src/modules/audio_device/include/fake_audio_device_impl.h:69

MethodSpeakerMute

src/modules/audio_device/include/fake_audio_device_impl.h:90

MethodSpeakerMuteIsAvailable

Speaker mute control

src/modules/audio_device/include/fake_audio_device_impl.h:88

MethodSpeakerVolume

src/modules/audio_device/include/fake_audio_device_impl.h:76

MethodSpeakerVolumeIsAvailable

Speaker volume controls

src/modules/audio_device/include/fake_audio_device_impl.h:74

src/audio/audio_receive_stream.cc:193

src/video/video_receive_stream2.cc:351

MethodStartAecDump

src/media/engine/webrtc_voice_engine.cc:679

MethodStartNextDecode

src/video/video_receive_stream2.cc:719

MethodStartPlayout

Audio transport control

src/modules/audio_device/include/fake_audio_device_impl.h:60

MethodStartRecording

src/modules/audio_device/include/fake_audio_device_impl.h:63

MethodStartWaitForNextFrameOnQueue

src/modules/video_coding/frame_buffer2.cc:105

MethodStatsCollector

src/pc/stats_collector.cc:539

MethodStereoPlayout

src/modules/audio_device/include/fake_audio_device_impl.h:100

MethodStereoPlayoutIsAvailable

src/modules/audio_device/include/fake_audio_device_impl.h:98

MethodStereoRecording

src/modules/audio_device/include/fake_audio_device_impl.h:103

MethodStereoRecordingIsAvailable

src/modules/audio_device/include/fake_audio_device_impl.h:101

src/audio/audio_receive_stream.cc:203

MethodStopAecDump

src/media/engine/webrtc_voice_engine.cc:695

MethodStopDecodeTimer

src/modules/video_coding/timing.cc:157

MethodStreamFinder

src/pc/channel.cc:51

MethodStreamStatisticianImpl

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:39

MethodStreamStatisticianLocked

src/modules/rtp_rtcp/source/receive_statistics_impl.h:121

MethodStreamSynchronization

src/video/stream_synchronization.cc:27

src/modules/rtp_rtcp/include/rtp_rtcp_defines.h:364

MethodSubtypeFromFormat

src/modules/audio_coding/neteq/decoder_database.cc:106

src/modules/rtp_rtcp/include/rtp_rtcp_defines.h:418

MethodSyncBufferSizeMs

src/modules/audio_coding/neteq/neteq_impl.cc:654

MethodTargetDelayInternal

src/modules/video_coding/timing.cc:295

MethodTargetDelayMs

src/modules/audio_coding/neteq/delay_manager.cc:272

MethodTargetDelayMs

src/modules/audio_coding/acm2/acm_receiver.cc:235

MethodTimeSinceFirstPacketInMs

src/modules/rtp_rtcp/include/rtp_rtcp_defines.h:376

MethodTimescaleAllowed

Checks if enough time has elapsed since the last successful timescale operation was done (i.e., accelerate or preemptive expand).

src/modules/audio_coding/neteq/decision_logic.h:156

src/media/base/media_channel.h:720

src/audio/audio_receive_stream.cc:36

MethodToStringMap

src/media/base/media_channel.h:733

MethodToStringMap

src/media/base/media_channel.h:750

MethodTotalBytes

src/modules/rtp_rtcp/include/rtp_rtcp_defines.h:338

MethodTransportFeedbackObserver

src/modules/rtp_rtcp/include/rtp_rtcp_defines.h:249

MethodUnderTargetLevel

src/modules/audio_coding/neteq/decision_logic.cc:424

MethodUnhandledPacketsBuffer

src/media/engine/unhandled_packets_buffer.cc:19

src/modules/audio_coding/neteq/neteq_impl.cc:2580

MethodUnregisterFromTransport

src/audio/audio_receive_stream.cc:162

MethodUnregisterFromTransport

src/video/video_receive_stream2.cc:329

src/modules/audio_coding/neteq/delay_manager.cc:140

MethodUpdateAllowedBitrateRange

src/media/engine/webrtc_voice_engine.cc:1107

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