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

MethodEncoderStreamFactory
TODO(bugs.webrtc.org/8785): Consider removing max_qp as member of EncoderStreamFactory and instead set this value individually for each stream in the
src/media/engine/webrtc_video_engine.cc:3522
MethodEqualsDisregardingFlexfec
src/media/engine/webrtc_video_engine.cc:3282
MethodExpandBufferSize
src/modules/video_coding/packet_buffer.cc:177
MethodExpectedPacketAvailable
src/modules/audio_coding/neteq/decision_logic.cc:306
MethodExtractBweInfo
src/pc/stats_collector.cc:1022
MethodExtractDataInfo
src/pc/stats_collector.cc:1275
MethodExtractMediaInfo
src/pc/stats_collector.cc:1167
MethodExtractPackets
src/modules/audio_coding/neteq/neteq_impl.cc:2127
MethodExtractSenderInfo
src/pc/stats_collector.cc:1242
MethodExtractSenderReceiverStats
src/pc/stats_collector.cc:1081
MethodExtractSessionInfo
src/pc/stats_collector.cc:856
MethodExtractSessionInfo_n
src/pc/stats_collector.cc:874
MethodExtractSessionInfo_s
src/pc/stats_collector.cc:930
MethodExtractStats
src/pc/stats_collector.cc:1140
MethodFakeAudioDeviceImpl
src/modules/audio_device/include/fake_audio_device_impl.h:17
MethodFillBitrateInfo
src/media/engine/webrtc_video_engine.cc:1688
MethodFillReceiverStats
src/media/engine/webrtc_video_engine.cc:1678
MethodFillSendAndReceiveCodecStats
src/media/engine/webrtc_video_engine.cc:1697
MethodFillSenderStats
src/media/engine/webrtc_video_engine.cc:1662
MethodFilterBufferLevel
src/modules/audio_coding/neteq/decision_logic.cc:238
MethodFindFrames
src/modules/video_coding/packet_buffer.cc:220
MethodFindMappingFor
src/media/engine/payload_type_mapper.cc:116
MethodFindNextFrame
src/modules/video_coding/frame_buffer2.cc:141
MethodFindReceiveStream
src/media/engine/webrtc_video_engine.cc:3435
MethodFrameBuffer
src/modules/video_coding/frame_buffer2.h:54
MethodFrameBuffer
src/modules/video_coding/frame_buffer2.cc:53
MethodFrameInfo
src/modules/video_coding/frame_buffer2.cc:734
MethodFreeFrame
src/rtd/src/rtd_api_impl.cpp:64
MethodFuturePacketAvailable
src/modules/audio_coding/neteq/decision_logic.cc:336
MethodGenerateKeyFrame
src/media/engine/webrtc_video_engine.cc:3253
MethodGenerateKeyFrame
src/video/video_receive_stream2.cc:1019
MethodGetANAStats
src/modules/audio_coding/acm2/audio_coding_module.cc:601
MethodGetAssociatedSendStreamForTesting
src/audio/audio_receive_stream.cc:470
MethodGetAudioFrameWithInfo
src/audio/channel_receive.cc:391
MethodGetAudioFrameWithInfo
src/audio/audio_receive_stream.cc:380
MethodGetAudioInternal
src/modules/audio_coding/neteq/neteq_impl.cc:956
MethodGetAudioState
src/media/engine/webrtc_voice_engine.cc:419
MethodGetBaseMinimumDelay
src/modules/audio_coding/neteq/decision_logic.h:97
MethodGetBaseMinimumDelayMs
src/modules/audio_coding/acm2/acm_receiver.cc:84
MethodGetBaseMinimumPlayoutDelayMs
src/media/engine/webrtc_voice_engine.cc:1300
MethodGetBaseMinimumPlayoutDelayMs
src/audio/channel_receive.cc:1022
MethodGetBaseMinimumPlayoutDelayMs
src/audio/audio_receive_stream.cc:370
MethodGetBaseMinimumPlayoutDelayMs
src/video/video_receive_stream2.cc:562
MethodGetChangedRecvParameters
src/media/engine/webrtc_video_engine.cc:1165
MethodGetChangedSendParameters
src/media/engine/webrtc_video_engine.cc:806
MethodGetCodecNameFromPayloadType
src/media/engine/webrtc_video_engine.cc:3136
MethodGetDecision
src/modules/audio_coding/neteq/neteq_impl.cc:1206
MethodGetDecision
src/modules/audio_coding/neteq/decision_logic.cc:106
MethodGetDecodingCallStatistics
src/modules/audio_coding/acm2/acm_receiver.cc:343
MethodGetDeduplicatedRtpHeaderExtensions
src/pc/channel.cc:756
MethodGetDefaultRtpReceiveParameters
src/media/engine/webrtc_voice_engine.cc:1545
MethodGetDefaultRtpReceiveParameters
src/media/engine/webrtc_video_engine.cc:1145
MethodGetDegradationPreference
src/media/engine/webrtc_video_engine.cc:2178
MethodGetFractionLostInPercent
src/modules/rtp_rtcp/source/receive_statistics_impl.h:131
MethodGetInfo
src/audio/audio_receive_stream.cc:404
MethodGetInfoFromConfigInternal
src/api/audio_codecs/audio_format.cc:114
MethodGetLastSliceQp
src/common_video/h264/h264_bitstream_parser.cc:334
MethodGetLatestDecodedAudioFrameTimestamp
src/modules/audio_coding/neteq/neteq_impl.cc:2325
MethodGetLatestDecodedAudioFrameTimestamp
src/modules/audio_coding/acm2/acm_receiver.cc:349
MethodGetLatestDecodedAudioFrameTimestamp
src/modules/audio_coding/acm2/audio_coding_module.cc:609
MethodGetLatestDecodedAudioFrameTimestamp
src/audio/audio_receive_stream.cc:482
MethodGetMappingFor
src/media/engine/payload_type_mapper.cc:96
MethodGetMaxWaitMs
src/video/video_receive_stream2.cc:714
MethodGetNackBatch
src/modules/video_coding/nack_requester.cc:333
MethodGetNackList
src/modules/audio_coding/acm2/acm_receiver.cc:323
MethodGetNetworkStatistics
TODO(turajs) change the return value to void. Also change the corresponding NetEq function.
src/modules/audio_coding/acm2/audio_coding_module.cc:588
MethodGetNextFrame
src/modules/video_coding/frame_buffer2.cc:251
MethodGetOrCreateStatistician
src/modules/rtp_rtcp/source/receive_statistics_impl.cc:367
MethodGetPeriodFractionLostInPercent
src/modules/rtp_rtcp/source/receive_statistics_impl.h:135
MethodGetPlayoutAudioParameters
src/modules/audio_device/include/fake_audio_device_impl.h:121
MethodGetPlayoutRtpTimestamp
src/audio/channel_receive.cc:982
MethodGetPlayoutRtpTimestamp
src/audio/audio_receive_stream.cc:411
MethodGetPlayoutTimestamp
src/modules/audio_coding/acm2/acm_receiver.cc:227
MethodGetRTT
src/audio/channel_receive.cc:1100
MethodGetReceiveStreamDataCounters
src/modules/rtp_rtcp/source/receive_statistics_impl.h:139
MethodGetRecordAudioParameters
src/modules/audio_device/include/fake_audio_device_impl.h:122
MethodGetReport
src/pc/stats_collector.cc:1297
MethodGetRtpHeaderExtensions
src/media/engine/webrtc_voice_engine.cc:663
MethodGetRtpHeaderExtensions
src/media/engine/webrtc_video_engine.cc:656
MethodGetRtpReceiveParameters
src/media/engine/webrtc_voice_engine.cc:1525
MethodGetRtpReceiveParameters
src/media/engine/webrtc_video_engine.cc:1123
MethodGetRtpSendParameters
src/media/engine/webrtc_voice_engine.cc:1449
MethodGetRtpSendParameters
src/media/engine/webrtc_video_engine.cc:1055
MethodGetRtpTimestampRateHz
src/audio/channel_receive.cc:1086
MethodGetSendCodec
src/media/engine/webrtc_video_engine.cc:1266
MethodGetSources
src/media/engine/webrtc_voice_engine.cc:1305
MethodGetSources
src/media/engine/webrtc_video_engine.cc:2016
MethodGetSources
src/audio/audio_receive_stream.cc:375
MethodGetSources
src/video/video_receive_stream2.cc:977
MethodGetStatistician
src/modules/rtp_rtcp/source/receive_statistics_impl.h:231
MethodGetStats
src/media/engine/webrtc_voice_engine.cc:961
MethodGetStats
src/media/engine/webrtc_voice_engine.cc:1260
MethodGetStats
src/modules/rtp_rtcp/source/receive_statistics_impl.h:127
MethodGetStats
src/modules/rtp_rtcp/source/receive_statistics_impl.cc:179
MethodGetStats
src/call/audio_receive_stream.h:170
MethodGetStats
src/video/video_receive_stream2.cc:508
MethodGetStatsOnWorkerThread
src/pc/stats_collector.cc:1136
MethodGetSupportedFormats
src/rtd/src/rtd_video_decoder_factory.cpp:57
MethodGetTimeNow
Wallclock time in ms.
src/pc/stats_collector.cc:552
MethodH264BitstreamParser
src/common_video/h264/h264_bitstream_parser.cc:41
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