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

⨍ Functions 1,316◇ Types & classes 335

↓ 2 callersFunctionExtractStatsFromList

src/pc/stats_collector.cc:469

↓ 2 callersFunctionFindCodec

src/media/engine/webrtc_voice_engine.cc:126

↓ 2 callersMethodFreeBuffer

src/rtd/src/rtd_frame_queue.cpp:50

↓ 2 callersMethodGetActiveCngDecoder

src/modules/audio_coding/neteq/decoder_database.cc:261

↓ 2 callersFunctionGetAudioNetworkAdaptorConfig

src/media/engine/webrtc_voice_engine.cc:151

↓ 2 callersMethodGetBaseMinimumPlayoutDelayMs

src/media/engine/webrtc_video_engine.cc:1987

↓ 2 callersMethodGetDecoderFormat

src/modules/audio_coding/neteq/neteq_impl.cc:591

↓ 2 callersMethodGetDefaultReceiveStreamSsrc

src/media/engine/webrtc_video_engine.cc:2004

↓ 2 callersMethodGetInfo

src/video/video_receive_stream2.cc:683

↓ 2 callersMethodGetInfoFromConfig

src/api/audio_codecs/audio_format.cc:71

↓ 2 callersMethodGetLifetimeStatistics

src/modules/audio_coding/neteq/neteq_impl.cc:536

↓ 2 callersMethodGetReceiveStreamDataCounters

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:286

↓ 2 callersMethodGetRtpParameters

src/media/engine/webrtc_video_engine.cc:2383

↓ 2 callersMethodGetStats

src/media/engine/webrtc_voice_engine.cc:2409

↓ 2 callersFunctionGetVp9LayersFromFieldTrialGroup

src/media/engine/webrtc_video_engine.cc:324

↓ 2 callersMethodInsertPacket

src/modules/video_coding/packet_buffer.cc:63

↓ 2 callersMethodIsCPresented

src/api/audio_codecs/audio_format.cc:57

↓ 2 callersFunctionIsDisabled

src/media/engine/webrtc_voice_engine.cc:213

↓ 2 callersFunctionIsDisabled

src/media/engine/webrtc_video_engine.cc:74

↓ 2 callersFunctionIsKeyFrameAndUnspecifiedResolution

src/video/video_receive_stream2.cc:184

↓ 2 callersMethodIsSupportedDecoder

src/rtd/src/rtd_audio_decoder_factory.cpp:101

↓ 2 callersFunctionIsTemporalLayersSupported

src/media/engine/webrtc_video_engine.cc:224

↓ 2 callersMethodLastDecoder

src/modules/audio_coding/acm2/acm_receiver.cc:239

↓ 2 callersMethodManageFrame

src/video/rtp_video_stream_receiver2.cc:947

↓ 2 callersFunctionMergeInfoAboutOutboundRtpSubstreams

src/media/engine/webrtc_video_engine.cc:406

↓ 2 callersFunctionNumActiveStreams

src/media/engine/webrtc_video_engine.cc:396

↓ 2 callersMethodOnCompleteFrame

src/video/video_receive_stream2.cc:638

↓ 2 callersMethodOnRtpPacket

src/pc/channel.cc:441

↓ 2 callersMethodOnSentPacket

src/pc/rtp_transport.cc:242

↓ 2 callersMethodPrepareReport

src/pc/stats_collector.cc:688

↓ 2 callersMethodReadFront

src/rtd/src/rtd_frame_queue.cpp:37

↓ 2 callersMethodRegisterAudioCallback

Full-duplex transportation of PCM audio

src/modules/audio_device/include/fake_audio_device_impl.h:24

↓ 2 callersMethodRegisterRtpDemuxerSink

src/pc/rtp_transport.cc:165

↓ 2 callersMethodRequestKeyFrame

src/video/video_receive_stream2.cc:627

↓ 2 callersMethodRequestKeyFrame

src/video/rtp_video_stream_receiver2.cc:141

↓ 2 callersFunctionResample

Resample audio in `frame` to given sample rate preserving the channel count and place the result in `destination`.

src/audio/audio_transport_impl.cc:69

↓ 2 callersMethodReset

src/modules/audio_coding/neteq/decision_logic.cc:74

↓ 2 callersMethodResetUnsignaledRecvStream

src/media/engine/webrtc_voice_engine.cc:2103

↓ 2 callersFunctionRtpSendParametersFromMediaDescription

src/pc/channel.cc:107

↓ 2 callersMethodSendNack

src/video/video_receive_stream2.cc:619

↓ 2 callersMethodSendNack

src/video/rtp_video_stream_receiver2.cc:146

↓ 2 callersMethodSetActualClockRate

src/modules/audio_coding/neteq/red_payload_splitter.h:49

↓ 2 callersMethodSetAll

src/media/base/media_channel.h:109

↓ 2 callersMethodSetAssociatedSendChannel

src/audio/channel_receive.cc:906

↓ 2 callersFunctionSetAudioProcessingStats

src/pc/stats_collector.cc:141

↓ 2 callersMethodSetBaseMinimumPlayoutDelayMs

src/media/engine/webrtc_video_engine.cc:1959

↓ 2 callersMethodSetDecoderMap

Set a new payload type -> decoder map.

src/media/engine/webrtc_voice_engine.cc:1255

↓ 2 callersMethodSetId

src/rtd/src/rtd_signaling.h:16

↓ 2 callersMethodSetInterface

src/media/engine/webrtc_video_engine.cc:1883

↓ 2 callersMethodSetLocalSsrc

src/media/engine/webrtc_video_engine.cc:2956

↓ 2 callersMethodSetMaximumDelay

src/modules/audio_coding/neteq/neteq_impl.cc:458

↓ 2 callersMethodSetMinimumDelay

src/modules/audio_coding/neteq/neteq_impl.cc:448

↓ 2 callersMethodSetNACKStatus

src/audio/channel_receive.cc:877

↓ 2 callersMethodSetOption

src/pc/channel.cc:327

↓ 2 callersMethodSetPlayout

src/media/engine/webrtc_voice_engine.cc:1909

↓ 2 callersMethodSetRawAudioSink

src/media/engine/webrtc_voice_engine.cc:2553

↓ 2 callersMethodSetReceiveCodecs

src/audio/channel_receive.cc:636

↓ 2 callersMethodSetRtcpOption

src/pc/rtp_transport.cc:43

↓ 2 callersMethodSetRtpOption

src/pc/rtp_transport.cc:39

↓ 2 callersMethodSetSampleRate

src/modules/audio_coding/neteq/decision_logic.cc:98

↓ 2 callersMethodSetSendParameters

src/media/engine/webrtc_voice_engine.cc:1367

↓ 2 callersMethodStop

src/video/video_receive_stream2.cc:426

↓ 2 callersFunctionSupportedH264Codecs

src/modules/video_coding/codecs/h264/h264.cc:69

↓ 2 callersMethodTargetVideoDelay

src/modules/video_coding/timing.cc:290

↓ 2 callersFunctionUpdateMeasurements

src/video/rtp_streams_synchronizer2.cc:31

↓ 2 callersMethodUpdateRtt

src/video/rtp_video_stream_receiver2.cc:923

↓ 2 callersFunctionValidateCodecFormats

src/media/engine/webrtc_video_engine.cc:242

↓ 2 callersFunctionValidateStreamParams

src/media/engine/webrtc_video_engine.cc:260

↓ 2 callersFunctionVectorToString

src/media/base/media_channel.h:88

↓ 2 callersFunctionVerifyUniquePayloadTypes

src/media/engine/webrtc_voice_engine.cc:140

↓ 2 callersMethodWriteBack

src/rtd/src/rtd_frame_queue.cpp:55

↓ 2 callersMethodarrival_time

Time in local time base as close as it can to packet arrived on the network.

src/modules/rtp_rtcp/source/rtp_packet_received.h:49

↓ 2 callersMethodcodec

src/video/video_receive_stream2.cc:97

↓ 2 callersMethodid

src/video/video_receive_stream2.cc:678

↓ 2 callersMethodlast_output_sample_rate_hz

src/modules/audio_coding/neteq/neteq_impl.cc:585

↓ 2 callersFunctionpthread_mutex_init

src/rtd/src/rtd_def.h:132

↓ 2 callersFunctionrtd_get_log_level

src/rtd/src/ffmpeg/rtd_dec.c:48

↓ 2 callersMethodset_payload_type_frequency

src/modules/rtp_rtcp/source/rtp_packet_received.h:66

↓ 2 callersFunctionset_stream_pts_info

src/rtd/src/ffmpeg/rtd_dec.c:356

↓ 1 callersFunctionAcmConfig

src/audio/channel_receive.cc:67

↓ 1 callersFunctionAdapterTypeToStatsType

src/pc/stats_collector.cc:513

↓ 1 callersFunctionAddDefaultFeedbackParams

src/media/engine/webrtc_video_engine.cc:93

↓ 1 callersMethodAddReceiveCodec

src/video/rtp_video_stream_receiver2.cc:328

↓ 1 callersMethodAddRecvStream

src/media/engine/webrtc_voice_engine.cc:2032

↓ 1 callersMethodAddRecvStream

src/media/engine/webrtc_video_engine.cc:1434

↓ 1 callersMethodAddSampleMemory

Adds `value` to `sample_memory_`.

src/modules/audio_coding/neteq/decision_logic.h:76

↓ 1 callersMethodAddSendStream

src/media/engine/webrtc_voice_engine.cc:1969

↓ 1 callersFunctionAddTrackReport

src/pc/stats_collector.cc:93

↓ 1 callersMethodAudioDecoderUninit

src/rtd/src/rtd_engine_impl.cpp:387

↓ 1 callersMethodBackfillBufferedPackets

src/media/engine/webrtc_video_engine.cc:1815

↓ 1 callersMethodBackfillPackets

Backfill `consumer` with all stored packet related `ssrcs`.

src/media/engine/unhandled_packets_buffer.cc:39

↓ 1 callersFunctionBuildReceiveStreamConfig

TODO(tommi): Constructing a receive stream could be made simpler. Move some of this boiler plate code into the config structs themselves.

src/media/engine/webrtc_voice_engine.cc:253

↓ 1 callersMethodBuiltInAECIsAvailable

Only supported on Android.

src/modules/audio_device/include/fake_audio_device_impl.h:109

↓ 1 callersMethodBuiltInAGCIsAvailable

src/modules/audio_device/include/fake_audio_device_impl.h:110

↓ 1 callersMethodBuiltInNSIsAvailable

src/modules/audio_device/include/fake_audio_device_impl.h:111

↓ 1 callersMethodCheckPayloadTypes

src/modules/audio_coding/neteq/decoder_database.cc:302

↓ 1 callersMethodCheckRedPayloads

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

↓ 1 callersMethodClear

src/media/base/media_channel.h:665

↓ 1 callersMethodClear

src/modules/video_coding/frame_buffer2.cc:377

↓ 1 callersMethodClearSource

Stops sending by setting the sink of the AudioSource to nullptr. No data callback will be received after this method. This method is called on the lib

src/media/engine/webrtc_voice_engine.cc:986

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