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

⨍ Functions 1,316◇ Types & classes 335

↓ 54 callersFunctionToString

Dumps an AudioCodec in RFC 2327-ish format.

src/media/engine/webrtc_voice_engine.cc:108

↓ 31 callersMethodToString

src/media/base/media_channel.h:122

↓ 24 callersMethodChannels

src/modules/audio_coding/codecs/aac/audio_decoder_aac.cc:122

↓ 23 callersMethodReadExponentialGolomb

src/rtc_base/bit_buffer.h:98

↓ 20 callersMethodReset

src/modules/audio_coding/neteq/delay_manager.cc:260

↓ 19 callersMethodReadBits

ABSL_DEPRECATED("")

src/rtc_base/bit_buffer.h:60

↓ 17 callersFunctionSafeSetError

src/pc/channel.cc:83

↓ 15 callersMethodid

src/audio/audio_receive_stream.cc:399

↓ 14 callersFunctionpthread_mutex_unlock

src/rtd/src/rtd_def.h:144

↓ 13 callersMethodIsComfortNoise

src/modules/audio_coding/neteq/decoder_database.cc:287

↓ 13 callersFunctionIsEnabled

src/media/engine/webrtc_video_engine.cc:69

↓ 13 callersMethodSampleRateHz

src/modules/audio_coding/codecs/aac/audio_decoder_aac.cc:118

↓ 12 callersFunctionIsCodec

src/media/engine/webrtc_voice_engine.cc:122

↓ 11 callersMethodRead

src/modules/audio_coding/neteq/neteq_impl.cc:2531

↓ 10 callersMethodRelease

src/video/video_receive_stream2.cc:179

↓ 8 callersMethodProcess

src/modules/audio_coding/neteq/accelerate.cc:19

↓ 8 callersMethodSsrc

src/audio/audio_receive_stream.cc:391

↓ 8 callersMethodTargetDelayMs

src/modules/audio_coding/neteq/neteq_impl.cc:481

↓ 7 callersMethodAdd

src/modules/rtp_rtcp/include/rtp_rtcp_defines.h:311

↓ 7 callersMethodGetDecoderInfo

src/modules/audio_coding/neteq/decoder_database.cc:201

↓ 7 callersMethodSize

src/modules/audio_coding/neteq/decoder_database.cc:123

↓ 7 callersMethodStart

src/rtd/src/rtd_api_impl.cpp:31

↓ 7 callersMethodToString

src/pc/channel.cc:152

↓ 7 callersMethodUpdate

src/modules/audio_coding/neteq/neteq_impl.cc:2587

↓ 7 callersMethodadd_ssrc

src/media/base/media_channel.h:335

↓ 7 callersMethodssrc

Utility accessor for clients that make the assumption only one ssrc exists per media. This will eventually go away. Call sites that compare this to ze

src/media/base/media_channel.h:359

↓ 6 callersMethodClear

src/modules/video_coding/packet_buffer.cc:151

↓ 6 callersMethodGetStats

src/pc/stats_collector.cc:614

↓ 6 callersFunctionIsEnabled

src/media/engine/webrtc_voice_engine.cc:208

↓ 6 callersMethodIsRed

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

↓ 6 callersMethodParse

src/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc:326

↓ 6 callersMethodStop

src/modules/video_coding/frame_buffer2.cc:367

↓ 6 callersMethodheight

src/modules/video_coding/packet_buffer.h:46

↓ 6 callersMethodwidth

src/modules/video_coding/packet_buffer.h:45

↓ 5 callersMethodGetDecoder

src/modules/audio_coding/neteq/decoder_database.cc:70

↓ 5 callersMethodGetStatistician

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:359

↓ 5 callersMethodSetSink

src/media/engine/webrtc_video_engine.cc:1603

↓ 5 callersMethodSetSource

Starts the sending by setting ourselves as a sink to the AudioSource to get data callbacks. This method is called on the libjingle worker thread. TODO

src/media/engine/webrtc_voice_engine.cc:971

↓ 5 callersFunctionpthread_mutex_lock

src/rtd/src/rtd_def.h:140

↓ 4 callersMethodAddPacket

Store packet in buffer.

src/media/engine/unhandled_packets_buffer.cc:26

↓ 4 callersMethodCopy

Mostly used by tests. Avoid creating copies if you can.

src/call/video_receive_stream.h:165

↓ 4 callersFunctionCreateH264Format

src/modules/video_coding/codecs/h264/h264.cc:50

↓ 4 callersFunctionGetPayloadTypesAndDefaultCodecs

src/media/engine/webrtc_video_engine.cc:124

↓ 4 callersMethodInit

src/modules/audio_coding/neteq/neteq_impl.cc:2514

↓ 4 callersMethodMatches

src/api/audio_codecs/audio_format.cc:52

↓ 4 callersMethodOnReceivedPacket

src/modules/video_coding/nack_requester.cc:176

↓ 4 callersMethodReset

src/modules/video_coding/timing.cc:52

↓ 4 callersFunctionRtpParametersFromMediaDescription

src/pc/channel.cc:90

↓ 4 callersMethodSetAndGetRecordingState

src/video/video_receive_stream2.cc:981

↓ 4 callersMethodSetMaxReorderingThreshold

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:170

↓ 4 callersMethodSetMinimumPlayoutDelay

src/video/video_receive_stream2.cc:707

↓ 4 callersMethodSetRecvParameters

src/media/engine/webrtc_voice_engine.cc:1413

↓ 4 callersMethodSetSend

src/media/engine/webrtc_voice_engine.cc:1922

↓ 4 callersMethodTargetLevelMs

src/modules/audio_coding/neteq/decision_logic.h:78

↓ 4 callersMethodadm

src/media/engine/webrtc_voice_engine.cc:706

↓ 4 callersFunctiongetNowTimeMicrosecond

src/rtc_base/timer.cc:17

↓ 4 callersMethodis_first_packet_in_frame

src/modules/video_coding/packet_buffer.h:48

↓ 4 callersMethodmuted

src/media/engine/webrtc_voice_engine.cc:956

↓ 4 callersMethodrecovered

Flag if packet was recovered via RTX or FEC.

src/modules/rtp_rtcp/source/rtp_packet_received.h:62

↓ 3 callersMethodClear

src/rtd/src/rtd_frame_queue.cpp:29

↓ 3 callersFunctionComputeSendBitrate

`max_send_bitrate_bps` is the bitrate from "b=" in SDP. `rtp_max_bitrate_bps` is the bitrate from RtpSender::SetParameters.

src/media/engine/webrtc_voice_engine.cc:176

↓ 3 callersMethodCreate

src/modules/audio_coding/neteq/accelerate.cc:103

↓ 3 callersFunctionCreateTrackReport

src/pc/stats_collector.cc:104

↓ 3 callersMethodEmpty

src/modules/audio_coding/neteq/decoder_database.cc:119

↓ 3 callersMethodEnable

src/pc/channel.cc:259

↓ 3 callersMethodFilteredCurrentDelayMs

src/modules/audio_coding/neteq/neteq_impl.cc:487

↓ 3 callersMethodGetActiveDecoder

src/modules/audio_coding/neteq/decoder_database.cc:236

↓ 3 callersMethodGetStats

src/media/engine/webrtc_video_engine.cc:1627

↓ 3 callersMethodInit

src/media/engine/webrtc_voice_engine.cc:339

↓ 3 callersMethodIsDefaultStream

src/media/engine/webrtc_video_engine.cc:3104

↓ 3 callersMethodIsDtmf

src/modules/audio_coding/neteq/decoder_database.cc:292

↓ 3 callersMethodIsDtxPacket

src/modules/audio_coding/codecs/aac/audio_decoder_aac.cc:26

↓ 3 callersFunctionIsH264CodecSupported

If H.264 OpenH264/FFmpeg codec is supported.

src/modules/video_coding/codecs/h264/h264.cc:39

↓ 3 callersFunctionIsLayerActive

src/media/engine/webrtc_video_engine.cc:379

↓ 3 callersMethodIsWritable

src/pc/rtp_transport.cc:122

↓ 3 callersMethodRenderTimeMs

src/modules/video_coding/timing.cc:176

↓ 3 callersMethodSendBufferedRtcpFeedback

src/video/rtp_video_stream_receiver2.cc:175

↓ 3 callersMethodSetExtmapAllowMixed

src/media/engine/webrtc_voice_engine.cc:879

↓ 3 callersMethodSetRtpExtensions

src/media/engine/webrtc_voice_engine.cc:872

↓ 3 callersMethodSize

src/rtd/src/rtd_frame_queue.cpp:24

↓ 3 callersMethodSubtract

src/modules/rtp_rtcp/include/rtp_rtcp_defines.h:318

↓ 3 callersFunctionToStringIfSet

src/media/base/media_channel.h:75

↓ 3 callersMethodis_last_packet_in_frame

src/modules/video_coding/packet_buffer.h:54

↓ 3 callersMethodnoise_fast_forward

src/modules/audio_coding/neteq/decision_logic.h:108

↓ 3 callersMethodpayload_type_frequency

src/modules/rtp_rtcp/source/rtp_packet_received.h:65

↓ 3 callersMethodset_prev_time_scale

src/modules/audio_coding/neteq/decision_logic.h:115

↓ 3 callersMethodset_sample_memory

Accessors and mutators.

src/modules/audio_coding/neteq/decision_logic.h:107

↓ 2 callersMethodApplyOptions

src/media/engine/webrtc_voice_engine.cc:435

↓ 2 callersFunctionAudioCodecToSdpAudioFormat

src/media/engine/payload_type_mapper.cc:21

↓ 2 callersMethodAudioDecoderInit

src/rtd/src/rtd_engine_impl.cpp:349

↓ 2 callersMethodBitrateReceived

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:291

↓ 2 callersMethodClearTo

src/modules/video_coding/packet_buffer.cc:115

↓ 2 callersMethodClose

src/rtd/src/rtd_demuxer.cpp:121

↓ 2 callersMethodCurrentDelayMs

src/video/video_receive_stream2.cc:1025

↓ 2 callersMethodDecode

src/modules/audio_coding/neteq/neteq_impl.cc:1499

↓ 2 callersFunctionDetermineMaxWaitForFrame

src/video/video_receive_stream2.cc:197

↓ 2 callersMethodEnableRetransmitDetection

src/modules/rtp_rtcp/source/receive_statistics_impl.cc:175

↓ 2 callersFunctionExtractCommonReceiveProperties

src/pc/stats_collector.cc:136

↓ 2 callersFunctionExtractCommonSendProperties

src/pc/stats_collector.cc:122

↓ 2 callersFunctionExtractStats

src/pc/stats_collector.cc:177

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