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

⨍ Functions 1,316◇ Types & classes 335

↓ 1 callersMethodClearUpTo

src/modules/video_coding/nack_requester.cc:252

↓ 1 callersMethodCngRfc3389On

These methods test the `cng_state_` for different conditions.

src/modules/audio_coding/neteq/decision_logic.h:63

↓ 1 callersFunctionCodecVectorToString

src/media/engine/webrtc_video_engine.cc:229

↓ 1 callersMethodCommand

src/rtd/src/rtd_demuxer.cpp:104

↓ 1 callersMethodComputeDelays

src/video/stream_synchronization.cc:64

↓ 1 callersMethodComputeRelativeDelay

src/video/stream_synchronization.cc:34

↓ 1 callersMethodConfigureSync

src/video/rtp_streams_synchronizer2.cc:56

↓ 1 callersMethodConnectAndWaitResponse

src/rtd/src/rtd_signaling.cpp:36

↓ 1 callersFunctionCreateChannelReceive

src/audio/audio_receive_stream.cc:70

↓ 1 callersFunctionCreateDecoderVideoCodec

src/video/video_receive_stream2.cc:116

↓ 1 callersFunctionCreateMediaChannelStatsGatherer

src/pc/stats_collector.cc:1152

↓ 1 callersFunctionCreateNetEq

src/modules/audio_coding/acm2/acm_receiver.cc:38

↓ 1 callersFunctionCreateNetEqController

src/modules/audio_coding/neteq/neteq_impl.cc:59

↓ 1 callersFunctionCreateRtpRtcpModule

src/video/rtp_video_stream_receiver2.cc:78

↓ 1 callersMethodCreateVideoDecoder

src/rtd/src/rtd_video_decoder_factory.cpp:64

↓ 1 callersMethodCurrentNetworkStatistics

src/modules/audio_coding/neteq/neteq_impl.cc:511

↓ 1 callersMethodDecode

src/video/video_receive_stream2.cc:165

↓ 1 callersMethodDecodeAudio

src/rtd/src/rtd_engine_impl.cpp:369

↓ 1 callersMethodDisableNack

src/modules/audio_coding/neteq/neteq_impl.cc:634

↓ 1 callersMethodDuration

src/modules/audio_coding/codecs/aac/audio_decoder_aac.cc:16

↓ 1 callersMethodEnableBuiltInAEC

Enables the built-in audio effects. Only supported on Android.

src/modules/audio_device/include/fake_audio_device_impl.h:114

↓ 1 callersMethodEnableBuiltInAGC

src/modules/audio_device/include/fake_audio_device_impl.h:115

↓ 1 callersMethodEnableBuiltInNS

src/modules/audio_device/include/fake_audio_device_impl.h:116

↓ 1 callersMethodEnableNack

src/modules/audio_coding/neteq/neteq_impl.cc:623

↓ 1 callersMethodEnableRetransmitDetection

src/modules/rtp_rtcp/source/receive_statistics_impl.h:244

↓ 1 callersMethodEquals

src/modules/audio_coding/neteq/neteq_impl.cc:2607

↓ 1 callersMethodExpandDecision

src/modules/audio_coding/neteq/decision_logic.cc:207

↓ 1 callersFunctionExtractRemoteStats

src/pc/stats_collector.cc:429

↓ 1 callersMethodExtractStats

src/pc/stats_collector.cc:1108

↓ 1 callersMethodFillBitrateInfo

src/pc/channel.cc:1013

↓ 1 callersMethodFilteredCurrentDelayMs

src/modules/audio_coding/acm2/acm_receiver.cc:231

↓ 1 callersFunctionFindRequiredActiveLayers

src/media/engine/webrtc_video_engine.cc:385

↓ 1 callersMethodFlushBuffers

src/modules/audio_coding/neteq/neteq_impl.cc:610

↓ 1 callersMethodFlushBuffers

src/modules/audio_coding/acm2/acm_receiver.cc:217

↓ 1 callersMethodForceSpsPpsIdrIsH264Keyframe

src/modules/video_coding/packet_buffer.cc:162

↓ 1 callersMethodFrameContinuous

src/video/rtp_video_stream_receiver2.cc:1065

↓ 1 callersMethodFrameDecoded

src/video/rtp_video_stream_receiver2.cc:1078

↓ 1 callersMethodFreeFrame

src/rtd/src/rtd_demuxer.cpp:130

↓ 1 callersMethodGetAggregatedVideoSenderInfo

src/media/engine/webrtc_video_engine.cc:2697

↓ 1 callersMethodGetAudio

src/modules/audio_coding/neteq/neteq_impl.cc:353

↓ 1 callersMethodGetAudio

src/modules/audio_coding/acm2/acm_receiver.cc:145

↓ 1 callersMethodGetBaseMinimumDelay

src/modules/audio_coding/neteq/delay_manager.cc:319

↓ 1 callersMethodGetBaseMinimumDelayMs

src/modules/audio_coding/neteq/neteq_impl.cc:476

↓ 1 callersMethodGetBaseMinimumPlayoutDelayMs

src/media/engine/webrtc_voice_engine.cc:2201

↓ 1 callersMethodGetBufferedFramesTimeMs

src/modules/video_coding/frame_buffer2.cc:713

↓ 1 callersMethodGetCurrentEstimatedPlayoutNtpTimestampMs

src/audio/channel_receive.cc:999

↓ 1 callersMethodGetDecodingCallStatistics

src/audio/channel_receive.cc:952

↓ 1 callersMethodGetDefaultSink

src/media/engine/webrtc_video_engine.cc:605

↓ 1 callersFunctionGetDelayChainLengthMs

src/modules/audio_coding/neteq/neteq_impl.cc:77

↓ 1 callersMethodGetDelayEstimate

src/audio/channel_receive.cc:959

↓ 1 callersMethodGetFilteredBufferLevel

src/modules/audio_coding/neteq/decision_logic.h:102

↓ 1 callersMethodGetFractionLostInPercent

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:254

↓ 1 callersMethodGetLatestDecodedAudioFrameTimestamp

src/audio/channel_receive.cc:1009

↓ 1 callersMethodGetLocalSsrc

src/audio/channel_receive.cc:938

↓ 1 callersFunctionGetMaxDefaultVideoBitrateKbps

The selected thresholds for QVGA and VGA corresponded to a QP around 10. The change in QP declined above the selected bitrates.

src/media/engine/webrtc_video_engine.cc:306

↓ 1 callersMethodGetNackList

src/modules/audio_coding/neteq/neteq_impl.cc:640

↓ 1 callersMethodGetNetworkStatistics

src/modules/audio_coding/acm2/acm_receiver.cc:249

↓ 1 callersMethodGetNetworkStatistics

src/audio/channel_receive.cc:944

↓ 1 callersMethodGetOperationsAndState

src/modules/audio_coding/neteq/neteq_impl.cc:541

↓ 1 callersMethodGetPerLayerVideoSenderInfos

src/media/engine/webrtc_video_engine.cc:2584

↓ 1 callersMethodGetPeriodFractionLostInPercent

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:268

↓ 1 callersMethodGetPlayoutRtpTimestamp

src/video/video_receive_stream2.cc:695

↓ 1 callersMethodGetPlayoutTimestamp

src/modules/audio_coding/neteq/neteq_impl.cc:567

↓ 1 callersMethodGetRTCPStatistics

src/audio/channel_receive.cc:812

↓ 1 callersMethodGetReceiveCodec

src/audio/channel_receive.cc:630

↓ 1 callersFunctionGetRtdApiFuncs

src/rtd/src/rtd_api.cpp:78

↓ 1 callersMethodGetRtpParameters

src/media/engine/webrtc_voice_engine.cc:1310

↓ 1 callersFunctionGetSendNackDelay

src/modules/video_coding/nack_requester.cc:36

↓ 1 callersMethodGetSources

src/media/engine/webrtc_voice_engine.cc:2579

↓ 1 callersMethodGetSpeechOutputLevelFullRange

src/audio/channel_receive.cc:774

↓ 1 callersMethodGetStats

src/audio/audio_receive_stream.cc:261

↓ 1 callersMethodGetStatsOnWorkerThread

src/pc/stats_collector.cc:1103

↓ 1 callersMethodGetStreamInfo

src/rtd/src/rtd_engine_impl.cpp:321

↓ 1 callersMethodGetStreamSyncOffsetInMs

TODO(https://bugs.webrtc.org/7065): Move RtpToNtpEstimator out of RtpStreamsSynchronizer and into respective receive stream to always populate the est

src/video/rtp_streams_synchronizer2.cc:184

↓ 1 callersMethodGetSupportedDecoders

src/rtd/src/rtd_audio_decoder_factory.cpp:83

↓ 1 callersMethodGetSupportedFormats

src/media/engine/internal_decoder_factory.cc:26

↓ 1 callersMethodGetSyncInfo

src/audio/channel_receive.cc:1026

↓ 1 callersMethodGetSyncInfo

src/video/rtp_video_stream_receiver2.cc:344

↓ 1 callersMethodGetTimingFrameInfo

src/modules/video_coding/timing.cc:322

↓ 1 callersMethodGetTimings

src/modules/video_coding/timing.cc:300

↓ 1 callersMethodGetTotalOutputDuration

src/audio/channel_receive.cc:784

↓ 1 callersMethodGetTotalOutputEnergy

src/audio/channel_receive.cc:779

↓ 1 callersFunctionGetTrackIdBySsrc

src/pc/stats_collector.cc:441

↓ 1 callersMethodGetUniqueFramesSeen

Returns number of different frames seen.

src/video/rtp_video_stream_receiver2.h:123

↓ 1 callersMethodGetVideoReceiverInfo

src/media/engine/webrtc_video_engine.cc:3147

↓ 1 callersFunctionGetVp9SpatialLayersFromFieldTrial

src/media/engine/webrtc_video_engine.cc:347

↓ 1 callersFunctionGetVp9TemporalLayersFromFieldTrial

src/media/engine/webrtc_video_engine.cc:357

↓ 1 callersMethodHasFixedBitrate

src/api/audio_codecs/audio_format.h:113

↓ 1 callersMethodHasRemoteAudio

src/pc/stats_collector.cc:1119

↓ 1 callersFunctionIceCandidateTypeToStatsType

src/pc/stats_collector.cc:496

↓ 1 callersMethodIncomingTimestamp

src/modules/video_coding/timing.cc:171

↓ 1 callersMethodInitRecording

src/modules/audio_device/include/fake_audio_device_impl.h:56

↓ 1 callersMethodInitialize

src/rtd/src/rtd_api_impl.cpp:20

↓ 1 callersFunctionInitializeCaptureFrame

We want to process at the lowest sample rate and channel count possible without losing information. Choose the lowest native rate at least equal to th

src/audio/audio_transport_impl.cc:32

↓ 1 callersMethodInsertEmptyPacket

src/modules/audio_coding/neteq/neteq_impl.cc:297

↓ 1 callersMethodInsertFrame

src/modules/video_coding/frame_buffer2.cc:414

↓ 1 callersMethodInsertPacket

src/modules/audio_coding/acm2/acm_receiver.cc:100

↓ 1 callersMethodInsertPadding

src/modules/video_coding/packet_buffer.cc:155

↓ 1 callersFunctionIsCodecDisabledForSimulcast

Returns true if the given codec is disallowed from doing simulcast.

src/media/engine/webrtc_video_engine.cc:296

↓ 1 callersMethodIsDecryptable

src/video/rtp_video_stream_receiver2.cc:723

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