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

⨍ Functions 1,316◇ Types & classes 335

↓ 1 callersMethodSetOptions

src/media/engine/webrtc_voice_engine.cc:1563

↓ 1 callersMethodSetOutputVolume

src/media/engine/webrtc_voice_engine.cc:2144

↓ 1 callersMethodSetPacketAudioLength

src/modules/audio_coding/neteq/delay_manager.cc:251

↓ 1 callersMethodSetProtectionMode

src/modules/video_coding/frame_buffer2.cc:361

↓ 1 callersMethodSetRtpParameters

src/media/engine/webrtc_voice_engine.cc:1034

↓ 1 callersMethodSetRtpParameters

src/media/engine/webrtc_video_engine.cc:2311

↓ 1 callersMethodSetSend

src/media/engine/webrtc_video_engine.cc:1276

↓ 1 callersMethodSetSendCodecSpec

src/media/engine/webrtc_voice_engine.cc:866

↓ 1 callersMethodSetSendParameters

src/media/engine/webrtc_video_engine.cc:879

↓ 1 callersMethodSetSourceTracker

src/audio/channel_receive.cc:526

↓ 1 callersMethodSetStereoChannelSwapping

src/audio/audio_transport_impl.cc:283

↓ 1 callersMethodSetTargetBufferingDelay

src/video/stream_synchronization.cc:173

↓ 1 callersMethodSetTimestampInterval

src/modules/audio_coding/neteq/red_payload_splitter.cc:196

↓ 1 callersMethodSetUseTransportCc

src/media/engine/webrtc_voice_engine.cc:1243

↓ 1 callersMethodSetUseTransportCcAndNackHistory

src/audio/audio_receive_stream.cc:232

↓ 1 callersMethodSoftReset

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

↓ 1 callersMethodSplitRed

The method loops through a list of packets {A, B, C, ...}. Each packet is split into its corresponding RED payloads, {A1, A2, ...}, which is temporari

src/modules/audio_coding/neteq/red_payload_splitter.cc:35

↓ 1 callersMethodStartPlayout

src/audio/channel_receive.cc:619

↓ 1 callersMethodStartReceive

src/video/rtp_video_stream_receiver2.cc:1104

↓ 1 callersMethodStop

src/rtd/src/rtd_api_impl.cpp:47

↓ 1 callersMethodStopPlayout

src/modules/audio_device/include/fake_audio_device_impl.h:61

↓ 1 callersMethodStopPlayout

src/audio/channel_receive.cc:624

↓ 1 callersMethodStopReceive

src/video/rtp_video_stream_receiver2.cc:1109

↓ 1 callersMethodStopRecording

src/modules/audio_device/include/fake_audio_device_impl.h:64

↓ 1 callersFunctionStreamTypeToString

src/media/engine/webrtc_video_engine.cc:56

↓ 1 callersMethodTerminate

src/modules/audio_device/include/fake_audio_device_impl.h:28

↓ 1 callersMethodToAudioCodec

src/media/engine/payload_type_mapper.cc:125

↓ 1 callersMethodUnregisterNackModule

src/modules/video_coding/nack_requester.cc:68

↓ 1 callersMethodUnregisterRtpDemuxerSink

src/pc/rtp_transport.cc:175

↓ 1 callersFunctionUpdateCodecTypeHistogram

Adds a codec usage sample to the histogram.

src/modules/audio_coding/acm2/audio_coding_module.cc:194

↓ 1 callersMethodUpdateCounters

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:113

↓ 1 callersMethodUpdateCurrentDelay

src/modules/video_coding/timing.cc:100

↓ 1 callersFunctionValidateStreamParams

src/media/engine/webrtc_voice_engine.cc:94

↓ 1 callersMethodWaitUntilNext

src/rtc_base/timer.cc:44

↓ 1 callersMethodapm

src/media/engine/webrtc_voice_engine.cc:712

↓ 1 callersMethodarrival_time_ms

ABSL_DEPRECATED("Use arrival_time() instead")

src/modules/rtp_rtcp/source/rtp_packet_received.h:53

↓ 1 callersMethodcodec

src/modules/video_coding/packet_buffer.h:44

↓ 1 callersFunctioninitialize

src/rtd/src/ffmpeg/rtd_dec.c:337

↓ 1 callersMethodis_real_first_packet_in_frame

src/modules/video_coding/packet_buffer.h:51

↓ 1 callersMethodlast_packet_sample_rate_hz

src/modules/audio_coding/acm2/acm_receiver.cc:88

↓ 1 callersFunctionpthread_cond_init

src/rtd/src/rtd_def.h:149

↓ 1 callersFunctionpthread_cond_signal

src/rtd/src/rtd_def.h:190

↓ 1 callersFunctionpthread_cond_timedwait

src/rtd/src/rtd_def.h:169

↓ 1 callersMethodrender_time_ms

src/video/video_receive_stream2.h:67

↓ 1 callersMethodrtcp_mux_enabled

src/pc/rtp_transport.h:51

↓ 1 callersFunctionrtd_audio_resample

src/rtd/src/ffmpeg/rtd_dec.c:155

↓ 1 callersFunctionrtd_read_close

src/rtd/src/ffmpeg/rtd_dec.c:415

↓ 1 callersFunctionrtd_stream_info_init

src/rtd/src/ffmpeg/rtd_dec.c:367

↓ 1 callersMethodset_arrival_time_ms

ABSL_DEPRECATED("Use set_arrival_time() instead")

src/modules/rtp_rtcp/source/rtp_packet_received.h:57

↓ 1 callersMethodset_max_playout_delay

src/modules/video_coding/timing.cc:79

↓ 1 callersMethodset_min_playout_delay

src/modules/video_coding/timing.cc:68

↓ 1 callersMethodset_render_delay

src/modules/video_coding/timing.cc:63

↓ 1 callersMethodstart

src/rtc_base/timer.cc:67

Method ClearRecordableEncodedFrameCallback

src/media/engine/webrtc_video_engine.cc:3241

Method ScopedNackPeriodicProcessorRegistration

src/modules/video_coding/nack_requester.cc:83

Method SetDepacketizerToDecoderFrameTransformer

src/media/engine/webrtc_video_engine.cc:3262

Method SetEncoderToPacketizerFrameTransformer

src/media/engine/webrtc_video_engine.cc:2763

Method SetRecordableEncodedFrameCallback

src/media/engine/webrtc_video_engine.cc:3228

Method VideoSendStreamParameters

src/media/engine/webrtc_video_engine.cc:2042

Method ~ScopedNackPeriodicProcessorRegistration

src/modules/video_coding/nack_requester.cc:90

src/modules/audio_coding/codecs/aac/audio_decoder_aac.cc:9

MethodAccelerate

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

MethodAccelerateFactory

src/modules/audio_coding/neteq/accelerate.h:73

MethodAcmReceiver

src/modules/audio_coding/acm2/acm_receiver.cc:51

MethodActiveAudioLayer

Retrieve the currently utilized audio layer

src/modules/audio_device/include/fake_audio_device_impl.h:21

MethodAdaptivePtimeConfig

src/media/engine/webrtc_voice_engine.cc:236

src/modules/rtp_rtcp/include/rtp_rtcp_defines.h:352

MethodAdd10MsData

Add 10MS of raw (PCM) audio data to the encoder.

src/modules/audio_coding/acm2/audio_coding_module.cc:335

MethodAdd10MsDataInternal

src/modules/audio_coding/acm2/audio_coding_module.cc:345

MethodAddCandidateReport

src/pc/stats_collector.cc:815

MethodAddCertificateReports

src/pc/stats_collector.cc:728

MethodAddConnectionInfoReport

src/pc/stats_collector.cc:756

MethodAddLocalAudioTrack

src/pc/stats_collector.cc:580

MethodAddPacketsToNack

src/modules/video_coding/nack_requester.cc:297

MethodAddRecvStream_w

src/pc/channel.cc:567

MethodAddSendStream

src/media/engine/webrtc_video_engine.cc:1337

MethodAddStream

Adds a MediaStream with tracks that can be used as a `selector` in a call to GetStats.

src/pc/stats_collector.cc:558

src/pc/stats_collector.cc:568

MethodAddTransceiver

src/rtd/src/rtd_engine_impl.cpp:231

MethodApplyChangedParams

src/media/engine/webrtc_video_engine.cc:979

MethodAssociateSendStream

src/audio/audio_receive_stream.cc:432

MethodAudioCodecInfo

src/api/audio_codecs/audio_format.cc:262

MethodAudioCodecInfo

src/api/audio_codecs/audio_format.h:98

MethodAudioCodingModule

src/modules/audio_coding/include/audio_coding_module.h:64

MethodAudioCodingModuleImpl

src/modules/audio_coding/acm2/audio_coding_module.cc:209

MethodAudioDecoder

src/api/audio_codecs/audio_decoder.h:38

MethodAudioDecoderAacExternal

src/rtd/src/rtd_audio_decoder_factory.cpp:36

MethodAudioDecoderAacImpl

src/modules/audio_coding/codecs/aac/audio_decoder_aac.cc:53

MethodAudioDecoderOpusExternal

src/rtd/src/rtd_audio_decoder_factory.cpp:13

MethodAudioReceiveStream

src/audio/audio_receive_stream.cc:90

MethodAudioReceiveStream

src/audio/audio_receive_stream.h:64

MethodAudioTransportImpl

src/audio/audio_transport_impl.h:40

MethodAudioTransportImpl

src/audio/audio_transport_impl.cc:87

MethodBackoffSettings

src/modules/video_coding/nack_requester.cc:107

MethodBaseChannel

src/pc/channel.cc:118

MethodBitStreamReader

src/modules/audio_coding/neteq/neteq_impl.h:361

MethodBitrateReceived

src/modules/rtp_rtcp/source/receive_statistics_impl.h:143

MethodCalcFirstAudioFrameDuration

src/rtd/src/rtd_engine_impl.cpp:343

MethodCalcFirstVideoFrameDuration

src/rtd/src/rtd_engine_impl.cpp:337

MethodCalcRtpTimestampInterval

src/modules/audio_coding/neteq/neteq_impl.cc:2336

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