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

⨍ Functions 1,316◇ Types & classes 335

↓ 1 callersMethodIsEmpty

src/media/engine/unhandled_packets_buffer.h:46

↓ 1 callersMethodIsReadyToSend

src/pc/rtp_transport.h:69

↓ 1 callersMethodIsRetransmissionsEnabled

src/video/rtp_video_stream_receiver2.cc:713

↓ 1 callersFunctionIsScaleFactorsPowerOfTwo

src/media/engine/webrtc_video_engine.cc:83

↓ 1 callersMethodIsUlpfecEnabled

src/video/rtp_video_stream_receiver2.cc:709

↓ 1 callersMethodLastReceivedKeyframePacketMs

src/video/rtp_video_stream_receiver2.cc:937

↓ 1 callersMethodLastReceivedPacketMs

src/video/rtp_video_stream_receiver2.cc:929

↓ 1 callersMethodMakeAudioDecoder

src/rtd/src/rtd_audio_decoder_factory.cpp:105

↓ 1 callersMethodMaxWaitingTime

src/modules/video_coding/timing.cc:265

↓ 1 callersMethodMaybeAppendReportBlockAndReset

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:198

↓ 1 callersMethodMaybeBackfillPackets

src/media/engine/unhandled_packets_buffer.cc:70

↓ 1 callersFunctionMaybeConstructNackModule

src/video/rtp_video_stream_receiver2.cc:107

↓ 1 callersMethodMaybeLog

src/modules/audio_coding/acm2/audio_coding_module.cc:201

↓ 1 callersFunctionMinPositive

Returns its smallest positive argument. If neither argument is positive, returns an arbitrary nonpositive value.

src/media/engine/webrtc_voice_engine.cc:164

↓ 1 callersFunctionMinPositive

Returns its smallest positive argument. If neither argument is positive, returns an arbitrary nonpositive value.

src/media/engine/webrtc_video_engine.cc:369

↓ 1 callersMethodNetworkStatistics

src/modules/audio_coding/neteq/neteq_impl.cc:498

↓ 1 callersMethodNextFrame

src/modules/video_coding/frame_buffer2.cc:83

↓ 1 callersMethodNotifyMutedState

src/modules/audio_coding/neteq/decision_logic.h:86

↓ 1 callersMethodOnAudioFrame

src/rtd/src/rtd_audio_decoder_factory.cpp:62

↓ 1 callersMethodOnAudioFrame

src/rtd/src/rtd_demuxer.cpp:142

↓ 1 callersMethodOnData

src/media/engine/webrtc_voice_engine.cc:88

↓ 1 callersMethodOnDemuxerCriteriaUpdateComplete

src/media/engine/webrtc_voice_engine.cc:2119

↓ 1 callersMethodOnDemuxerCriteriaUpdatePending

Not implemented. TODO(https://crbug.com/webrtc/12676): Implement a fix for the unsignalled SSRC race that can happen when an m= section goes from rece

src/media/engine/webrtc_voice_engine.cc:2118

↓ 1 callersMethodOnEncodedImage

EncodedImageCallback implementation

src/rtd/src/rtd_engine_impl.cpp:491

↓ 1 callersMethodOnLocalSsrcChange

src/audio/channel_receive.cc:932

↓ 1 callersMethodOnPacketReceived

src/media/engine/webrtc_voice_engine.cc:2268

↓ 1 callersMethodOnPacketSent

src/media/engine/webrtc_voice_engine.cc:2338

↓ 1 callersMethodOnReadyToSend

src/pc/rtp_transport.cc:226

↓ 1 callersMethodOnReceivedFirstSequence

src/modules/video_coding/nack_requester.cc:269

↓ 1 callersMethodOnUnsignalledSsrc

src/media/engine/webrtc_video_engine.cc:573

↓ 1 callersMethodOnVideoFrame

src/rtd/src/rtd_demuxer.cpp:165

↓ 1 callersMethodOpen

src/rtd/src/rtd_demuxer.cpp:41

↓ 1 callersMethodPacketArrived

src/modules/audio_coding/neteq/decision_logic.cc:215

↓ 1 callersFunctionPacketBufferMaxSize

src/video/rtp_video_stream_receiver2.cc:61

↓ 1 callersMethodPacketDuration

src/modules/audio_coding/codecs/aac/audio_decoder_aac.cc:102

↓ 1 callersMethodPacketDurationRedundant

src/modules/audio_coding/codecs/aac/audio_decoder_aac.cc:106

↓ 1 callersMethodPaddingReceived

src/modules/video_coding/rtp_seq_num_only_ref_finder.cc:163

↓ 1 callersFunctionParseFuaNalu

src/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc:258

↓ 1 callersMethodParsePayload

src/modules/audio_coding/codecs/aac/audio_decoder_aac.cc:94

↓ 1 callersFunctionParseStapAStartOffsets

TODO(pbos): Avoid parsing this here as well as inside the jitter buffer.

src/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc:47

↓ 1 callersMethodPeakFound

src/modules/audio_coding/neteq/decision_logic.h:100

↓ 1 callersMethodPlayoutDelay

src/modules/audio_device/include/fake_audio_device_impl.h:106

↓ 1 callersFunctionPowerOfTwo

src/media/engine/webrtc_video_engine.cc:79

↓ 1 callersMethodPrepareADMReport

src/pc/stats_collector.cc:715

↓ 1 callersFunctionProcessCaptureFrame

src/audio/audio_transport_impl.cc:48

↓ 1 callersMethodProcessNacks

src/modules/video_coding/nack_requester.cc:166

↓ 1 callersFunctionProcessStapAOrSingleNalu

src/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc:69

↓ 1 callersMethodReadFrame

src/rtd/src/rtd_demuxer.cpp:57

↓ 1 callersMethodReadSignedExponentialGolomb

src/rtc_base/bit_buffer.h:106

↓ 1 callersMethodReceivedRTCPPacket

src/audio/channel_receive.cc:734

↓ 1 callersMethodRecording

src/modules/audio_device/include/fake_audio_device_impl.h:65

↓ 1 callersMethodRecordingIsInitialized

src/modules/audio_device/include/fake_audio_device_impl.h:57

↓ 1 callersMethodReduceAudioDelay

src/video/stream_synchronization.cc:188

↓ 1 callersMethodReduceVideoDelay

src/video/stream_synchronization.cc:192

↓ 1 callersMethodRegisterEmptyPacket

src/modules/audio_coding/neteq/decision_logic.h:84

↓ 1 callersMethodRegisterNackModule

src/modules/video_coding/nack_requester.cc:55

↓ 1 callersMethodRegisterPayload

src/modules/audio_coding/neteq/decoder_database.cc:166

↓ 1 callersMethodRegisterReceiverCongestionControlObjects

src/audio/channel_receive.cc:795

↓ 1 callersMethodRemove

src/modules/audio_coding/neteq/decoder_database.cc:181

↓ 1 callersMethodRemoveAll

src/modules/audio_coding/neteq/decoder_database.cc:195

↓ 1 callersMethodRemoveAllCodecs

src/modules/audio_coding/acm2/acm_receiver.cc:221

↓ 1 callersMethodRemoveAllPayloadTypes

src/modules/audio_coding/neteq/neteq_impl.cc:443

↓ 1 callersMethodRemoveRecvStream

src/media/engine/webrtc_voice_engine.cc:2083

↓ 1 callersMethodRemoveRecvStream

src/media/engine/webrtc_video_engine.cc:1553

↓ 1 callersMethodRemoveSendStream

src/media/engine/webrtc_voice_engine.cc:2006

↓ 1 callersMethodRequestPacketRetransmit

src/video/rtp_video_stream_receiver2.cc:717

↓ 1 callersMethodResetInitialDelay

src/modules/audio_coding/acm2/acm_receiver.cc:328

↓ 1 callersMethodResetReceiverCongestionControlObjects

src/audio/channel_receive.cc:805

↓ 1 callersMethodSendData

src/modules/audio_coding/include/audio_coding_module.h:41

↓ 1 callersMethodSendLossNotification

src/video/rtp_video_stream_receiver2.cc:161

↓ 1 callersMethodSendProcessedData

src/audio/audio_transport_impl.cc:189

↓ 1 callersMethodSendRtcpPacket

src/pc/rtp_transport.cc:135

↓ 1 callersMethodSendRtpPacket

src/pc/rtp_transport.cc:129

↓ 1 callersMethodSendTelephoneEvent

src/media/engine/webrtc_voice_engine.cc:933

↓ 1 callersMethodSetActiveDecoder

src/modules/audio_coding/neteq/decoder_database.cc:211

↓ 1 callersFunctionSetAudioFrameActivityAndType

src/modules/audio_coding/neteq/neteq_impl.cc:306

↓ 1 callersMethodSetAudioNetworkAdaptorConfig

src/media/engine/webrtc_voice_engine.cc:900

↓ 1 callersMethodSetBaseMinimumDelay

src/modules/audio_coding/neteq/delay_manager.cc:309

↓ 1 callersMethodSetBaseMinimumDelayMs

src/modules/audio_coding/neteq/neteq_impl.cc:468

↓ 1 callersMethodSetBaseMinimumPlayoutDelayMs

src/media/engine/webrtc_voice_engine.cc:2178

↓ 1 callersMethodSetChannelOutputVolumeScaling

src/audio/channel_receive.cc:789

↓ 1 callersMethodSetCngOff

Resets the `cng_state_` to kCngOff.

src/modules/audio_coding/neteq/decision_logic.h:67

↓ 1 callersMethodSetCodecs

src/modules/audio_coding/neteq/neteq_impl.cc:413

↓ 1 callersMethodSetCodecs

src/modules/audio_coding/acm2/acm_receiver.cc:213

↓ 1 callersMethodSetDepacketizerToDecoderFrameTransformer

src/media/engine/webrtc_voice_engine.cc:2604

↓ 1 callersMethodSetDepacketizerToDecoderFrameTransformer

src/media/engine/webrtc_video_engine.cc:3500

↓ 1 callersMethodSetEstimatedPlayoutNtpTimestampMs

src/video/video_receive_stream2.cc:701

↓ 1 callersMethodSetFeedbackParameters

src/media/engine/webrtc_video_engine.cc:2977

↓ 1 callersMethodSetFrameDecryptor

src/media/engine/webrtc_voice_engine.cc:2220

↓ 1 callersMethodSetFrameDecryptor

src/video/rtp_video_stream_receiver2.cc:901

↓ 1 callersMethodSetGain

src/audio/audio_receive_stream.cc:360

↓ 1 callersMethodSetJitterDelay

src/modules/video_coding/timing.cc:89

↓ 1 callersMethodSetLastDecodeScheduledTimestamp

src/modules/video_coding/timing.cc:183

↓ 1 callersMethodSetLatestDecodedAudioTimestamp

src/modules/video_coding/timing.cc:189

↓ 1 callersMethodSetLocalSsrc

src/audio/audio_receive_stream.cc:452

↓ 1 callersMethodSetMaxCompositionDelayInFrames

src/modules/video_coding/timing.cc:327

↓ 1 callersMethodSetMaxReorderingThreshold

src/modules/rtp_rtcp/source/receive_statistics_impl.h:235

↓ 1 callersMethodSetMid

src/media/engine/webrtc_voice_engine.cc:884

↓ 1 callersMethodSetMinimumDelay

src/modules/audio_coding/acm2/acm_receiver.cc:66

↓ 1 callersMethodSetMuted

src/media/engine/webrtc_voice_engine.cc:949

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