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

⨍ Functions 1,316◇ Types & classes 335

MethodOnLogMessage

src/rtd/src/rtd_log.cpp:13

MethodOnNetworkRouteChanged

src/pc/rtp_transport.cc:230

MethodOnNetworkRouteChanged

src/pc/channel.cc:354

MethodOnNetworkRouteChanged

src/media/engine/webrtc_voice_engine.cc:2349

MethodOnNetworkRouteChanged

src/media/engine/webrtc_video_engine.cc:1869

MethodOnPacketReceived

src/media/engine/webrtc_video_engine.cc:1711

MethodOnPacketSent

src/media/engine/webrtc_video_engine.cc:1803

MethodOnReadPacket

src/pc/rtp_transport.cc:259

MethodOnReadyToSend

src/media/engine/webrtc_voice_engine.cc:2401

MethodOnReadyToSend

src/media/engine/webrtc_video_engine.cc:1861

MethodOnReceivedPayloadData

src/audio/channel_receive.cc:316

MethodOnReceivedPayloadData

src/video/rtp_video_stream_receiver2.cc:471

MethodOnRecoveredPacket

src/video/rtp_video_stream_receiver2.cc:642

MethodOnRtcpPacketReceived

src/pc/rtp_transport.cc:254

MethodOnRtpPacket

src/modules/rtp_rtcp/source/receive_statistics_impl.h:227

MethodOnRtpPacket

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:351

MethodOnRtpPacket

src/audio/channel_receive.cc:646

MethodOnRtpPacket

This method handles both regular RTP packets and packets recovered via FlexFEC.

src/video/rtp_video_stream_receiver2.cc:667

MethodOnRtpPacketReceived

src/pc/rtp_transport.cc:249

MethodOnRttUpdate

src/video/video_receive_stream2.cc:670

MethodOnSdpOffer

src/rtd/src/rtd_engine_impl.cpp:395

MethodOnSignalingChange

PeerConnectionObserver implementation

src/rtd/src/rtd_engine_impl.cpp:408

MethodOnSuccess

src/rtd/src/rtd_engine_impl.cpp:32

MethodOnSuccess

src/rtd/src/rtd_engine_impl.cpp:50

MethodOnSuccess

src/rtd/src/rtd_engine_impl.cpp:69

MethodOnTransportReadyToSend

src/pc/channel.cc:377

MethodOnWritableState

src/pc/rtp_transport.cc:235

MethodOnWritableState

src/pc/channel.cc:345

src/rtd/src/rtd_engine_impl.cpp:124

src/modules/video_coding/packet_buffer.cc:37

src/modules/video_coding/packet_buffer.h:35

MethodPacketBuffer

src/modules/video_coding/packet_buffer.cc:46

MethodPacketTooEarly

src/modules/audio_coding/neteq/decision_logic.cc:434

MethodParseAndHandleEncapsulatingHeader

RTC_RUN_ON(packet_sequence_checker_)

src/video/rtp_video_stream_receiver2.cc:984

MethodParseBitstream

src/common_video/h264/h264_bitstream_parser.cc:325

MethodParseFromFieldTrials

src/modules/video_coding/nack_requester.cc:112

MethodParseGenericDependenciesExtension

RTC_RUN_ON(packet_sequence_checker_)

src/video/rtp_video_stream_receiver2.cc:366

MethodParseNonParameterSetNalu

src/common_video/h264/h264_bitstream_parser.cc:68

MethodParseSlice

src/common_video/h264/h264_bitstream_parser.cc:296

MethodParseStreamInfo

src/rtd/src/rtd_engine_impl.cpp:305

src/media/engine/webrtc_voice_engine.cc:228

src/modules/audio_coding/neteq/delay_manager.cc:47

MethodPayloadTypeMapper

src/media/engine/payload_type_mapper.cc:25

src/rtc_base/bit_buffer.h:69

src/modules/audio_device/include/fake_audio_device_impl.h:62

MethodPlayoutData10Ms

Get 10 milliseconds of raw audio data to play out. Automatic resample to the requested frequency.

src/modules/audio_coding/acm2/audio_coding_module.cc:571

MethodPlayoutDeviceName

src/modules/audio_device/include/fake_audio_device_impl.h:34

MethodPlayoutDevices

Device enumeration

src/modules/audio_device/include/fake_audio_device_impl.h:32

MethodPlayoutIsAvailable

Audio transport initialization

src/modules/audio_device/include/fake_audio_device_impl.h:52

MethodPlayoutIsInitialized

src/modules/audio_device/include/fake_audio_device_impl.h:54

MethodPotentialNewFrame

src/modules/video_coding/packet_buffer.cc:196

MethodPreemptiveExpand

src/modules/audio_coding/neteq/preemptive_expand.h:31

MethodPreemptiveExpandFactory

src/modules/audio_coding/neteq/preemptive_expand.h:77

MethodPreferredSampleRate

src/audio/channel_receive.cc:519

MethodPreferredSampleRate

src/audio/audio_receive_stream.cc:395

MethodPreprocessToAddData

Perform a resampling and down-mix if required. We down-mix only if encoder is mono and input is stereo. In case of dual-streaming, both encoders has t

src/modules/audio_coding/acm2/audio_coding_module.cc:422

src/modules/audio_coding/neteq/preemptive_expand.cc:21

MethodProcessNackModules

RTC_RUN_ON(sequence_)

src/modules/video_coding/nack_requester.cc:78

MethodPropagateContinuity

src/modules/video_coding/frame_buffer2.cc:522

MethodPropagateDecodability

src/modules/video_coding/frame_buffer2.cc:556

MethodProxySink

src/media/engine/webrtc_voice_engine.cc:84

MethodPullRenderData

Used by Chromium - same as NeedMorePlayData() but because Chrome has its own APM instance, does not call audio_processing_->ProcessReverseStream().

src/audio/audio_transport_impl.cc:248

MethodRTCPReportBlock

src/modules/rtp_rtcp/include/rtp_rtcp_defines.h:116

MethodRTC_GUARDED_BY

src/modules/audio_coding/neteq/neteq_impl.h:480

MethodRTC_GUARDED_BY

src/audio/channel_receive.cc:304

MethodRTC_GUARDED_BY

src/video/video_receive_stream2.h:261

MethodReadFrame

src/rtd/src/rtd_api_impl.cpp:57

MethodReadNonSymmetric

src/rtc_base/bit_buffer.h:85

MethodReadUInt16

src/rtc_base/bit_buffer.h:48

MethodReadUInt32

src/rtc_base/bit_buffer.h:51

MethodReadUInt8

src/rtc_base/bit_buffer.h:45

MethodReceivePacket

src/audio/channel_receive.cc:683

MethodReceivePacket

RTC_RUN_ON(packet_sequence_checker_)

src/video/rtp_video_stream_receiver2.cc:954

MethodReceiveStatisticsImpl

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:340

MethodReceiveStatisticsLocked

src/modules/rtp_rtcp/source/receive_statistics_impl.h:215

MethodReceivedRtpPacket

Checks if this StreamStatistician received any rtp packets.

src/modules/rtp_rtcp/source/receive_statistics_impl.h:75

MethodReconfigureAudioSendStream

src/media/engine/webrtc_voice_engine.cc:1182

MethodReconfigureEncoder

src/media/engine/webrtc_video_engine.cc:2552

MethodReconfigureForTesting

src/audio/audio_receive_stream.cc:167

MethodRecordedDataIsAvailable

Not used in Chromium. Process captured audio and distribute to all sending streams, and try to do this at the lowest possible sample rate.

src/audio/audio_transport_impl.cc:120

MethodRecordingDeviceName

src/modules/audio_device/include/fake_audio_device_impl.h:39

MethodRecordingDevices

src/modules/audio_device/include/fake_audio_device_impl.h:33

MethodRecordingIsAvailable

src/modules/audio_device/include/fake_audio_device_impl.h:55

MethodRecordingState

src/call/video_receive_stream.h:46

MethodRecreateWebRtcStream

src/media/engine/webrtc_video_engine.cc:2773

MethodRecreateWebRtcVideoStream

src/media/engine/webrtc_video_engine.cc:3044

MethodRedPayloadSplitter

src/modules/audio_coding/neteq/red_payload_splitter.h:29

MethodRegisterDecodeCompleteCallback

src/rtd/src/rtd_video_decoder_factory.cpp:40

MethodRegisterDecodeCompleteCallback

src/video/video_receive_stream2.cc:172

MethodRegisterPayloadType

src/modules/audio_coding/neteq/neteq_impl.cc:422

MethodRegisterRtpDemuxerSink_w

src/pc/channel.cc:487

MethodRegisterTransportCallback

Register a transport callback which will be called to deliver the encoded buffers.

src/modules/audio_coding/acm2/audio_coding_module.cc:327

MethodRegisterWithTransport

src/audio/audio_receive_stream.cc:154

MethodRegisterWithTransport

src/video/video_receive_stream2.cc:313

MethodReinitAfterExpands

src/modules/audio_coding/neteq/decision_logic.cc:429

src/rtd/src/rtd_video_decoder_factory.cpp:44

MethodRemoveLocalAudioTrack

src/pc/stats_collector.cc:602

MethodRemovePacketsUntilKeyFrame

src/modules/video_coding/nack_requester.cc:278

MethodRemovePayloadType

src/modules/audio_coding/neteq/neteq_impl.cc:432

MethodRemoveRecvStream_w

src/pc/channel.cc:571

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