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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 58 callersMethodReadUInt8
rtc_base/bit_buffer.cc:86
↓ 58 callersMethodSetSsrc
modules/rtp_rtcp/source/rtp_packet.cc:156
↓ 58 callersFunctionToUnsigned
logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h:36
↓ 58 callersMethodclose
()
sdk/android/api/org/webrtc/FileVideoCapturer.java:27
↓ 58 callersMethoddisconnect
Disconnects the client if not already disconnected. This will fire the onTCPClose event.
examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java:90
↓ 57 callersMethodCurrentTime
modules/pacing/pacing_controller.cc:186
↓ 57 callersMethodGetStats
media/base/video_adapter_unittest.cc:104
↓ 57 callersFunctionInt64MsToQ32x32
Converts |int64_t| milliseconds to Q32.32-formatted fixed-point seconds. Performs clamping if the result overflows or underflows.
system_wrappers/include/ntp_time.h:67
↓ 57 callersFunctionInt64MsToUQ32x32
Converts |int64_t| milliseconds to UQ32.32-formatted fixed-point seconds. Performs clamping if the result overflows or underflows.
system_wrappers/include/ntp_time.h:86
↓ 57 callersMethodSetAnswer
pc/srtp_filter.cc:66
↓ 57 callersMethodSetRTCPStatus
modules/rtp_rtcp/source/rtcp_sender.cc:212
↓ 57 callersMethodStrideU
api/video/i010_buffer.cc:182
↓ 57 callersMethodStrideV
api/video/i010_buffer.cc:185
↓ 57 callersMethodUpdate
modules/audio_processing/aec3/aec_state.cc:158
↓ 57 callersMethodWriteUInt8
rtc_base/bit_buffer.cc:251
↓ 57 callersMethodbegin
logging/rtc_event_log/rtc_event_log_parser.h:172
↓ 57 callersMethodwrite
(self, contents)
tools_webrtc/mb/mb_unittest.py:105
↓ 56 callersMethodAddStream
pc/peer_connection.cc:1445
↓ 56 callersMethodGetBitrate
audio/channel_send.cc:627
↓ 56 callersMethodIsPlusInfinity
rtc_base/units/unit_base.h:48
↓ 56 callersMethodIsZero
rtc_base/units/unit_base.h:43
↓ 56 callersFunctionMakeArrayView
api/array_view.h:284
↓ 56 callersMethodPost
rtc_base/thread.cc:484
↓ 56 callersMethodfactory
media/engine/simulcast_encoder_adapter_unittest.cc:396
↓ 55 callersFunctionABSL_FLAG
Flag for payload type.
video/video_replay.cc:48
↓ 55 callersFunctionErrorToString
modules/audio_device/win/core_audio_utility_win.cc:1516
↓ 55 callersMethodRun
rtc_base/thread.cc:808
↓ 55 callersMethodRunFor
test/scenario/scenario.cc:272
↓ 55 callersMethodSendRTCP
modules/rtp_rtcp/source/rtcp_sender.cc:696
↓ 55 callersMethodStop
pc/rtp_sender.cc:302
↓ 55 callersMethodWait
video/video_analyzer.cc:343
↓ 55 callersMethodrequests
Gets the number of NACKed RTP packets.
modules/rtp_rtcp/source/rtcp_nack_stats.h:28
↓ 54 callersMethodAppendTimeSeries
rtc_tools/rtc_event_log_visualizer/plot_base.cc:82
↓ 54 callersFunctionCreateRtpPacket
modules/rtp_rtcp/source/receive_statistics_unittest.cc:35
↓ 54 callersMethodGetRenderBuffer
modules/audio_processing/aec3/render_delay_buffer.cc:60
↓ 54 callersMethodRemoveTrack
pc/media_stream.cc:49
↓ 54 callersFunctionSamePacketAs
call/rtcp_demuxer_unittest.cc:101
↓ 54 callersMethodWriteBits
rtc_base/bit_buffer.cc:263
↓ 54 callersMethodbegin
rtc_base/buffer.h:194
↓ 54 callersMethodrelease
()
sdk/android/src/java/org/webrtc/MediaCodecWrapper.java:34
↓ 54 callersMethodset_packet_type
modules/rtp_rtcp/source/rtp_packet_to_send.h:51
↓ 54 callersMethodwrite_state
p2p/base/connection.h:112
↓ 53 callersMethodAddInternalSocket
p2p/base/turn_server.cc:146
↓ 53 callersMethodAddVideoConfig
test/pc/e2e/peer_connection_quality_test.h:121
↓ 53 callersMethodAllocate
modules/video_coding/codecs/vp9/svc_rate_allocator.cc:198
↓ 53 callersMethodCreateConnection
p2p/base/port_unittest.cc:199
↓ 53 callersMethodRegisterRtpHeaderExtension
modules/rtp_rtcp/source/rtp_sender.cc:163
↓ 53 callersMethodSetSend
media/engine/webrtc_video_engine_unittest.cc:1553
↓ 53 callersMethodwritable
p2p/base/dtls_transport.cc:451
↓ 52 callersMethodCreateClient
test/scenario/scenario.cc:108
↓ 52 callersMethodas_audio
Try to cast this media description to an AudioContentDescription. Returns nullptr if the cast fails.
pc/session_description.h:72
↓ 52 callersMethodheight
common_video/video_frame_buffer.cc:48
↓ 52 callersMethodrtcp_mux
pc/session_description.h:109
↓ 52 callersMethodtoString
()
rtc_base/java/src/org/webrtc/Size.java:26
↓ 52 callersMethodwidth
common_video/video_frame_buffer.cc:46
↓ 51 callersMethodAddContentName
pc/session_description.cc:75
↓ 51 callersMethodAddStats
stats/rtc_stats_report.cc:76
↓ 51 callersFunctionGetChannelData
audio/utility/audio_frame_operations_unittest.cc:90
↓ 51 callersMethodGetTransportInfoByName
pc/session_description.cc:211
↓ 51 callersFunctionInjectAfter
Add some extra |newlines| to the |message| after |line|.
pc/webrtc_sdp_unittest.cc:929
↓ 51 callersMethodRegisterPlot
rtc_tools/rtc_event_log_visualizer/main.cc:131
↓ 51 callersMethodequals
equals() checks sdpMid, sdpMLineIndex, and sdp for equality.
sdk/android/api/org/webrtc/IceCandidate.java:63
↓ 51 callersMethodlog_time_ms
logging/rtc_event_log/logged_events.h:264
↓ 50 callersFunctionAddAudioVideoSections
pc/media_session_unittest.cc:330
↓ 50 callersFunctionContains
system_wrappers/source/rtp_to_ntp_estimator.cc:32
↓ 50 callersFunctionCurrentThreadId
rtc_base/platform_thread_types.cc:20
↓ 50 callersMethodGet
modules/rtp_rtcp/source/rtp_sequence_number_map.cc:97
↓ 50 callersMethodGetStatsReport
pc/rtc_stats_collector_unittest.cc:368
↓ 50 callersMethodPing
p2p/base/port_unittest.cc:308
↓ 50 callersFunctionSafeClamp
rtc_base/numerics/safe_minmax.h:320
↓ 50 callersFunctionWebRtcSpl_MemSetW16
common_audio/signal_processing/copy_set_operations.c:29
↓ 50 callersMethodWriteExponentialGolomb
rtc_base/bit_buffer.cc:329
↓ 50 callersMethodload
Loads a native library with the given name. @return True on success
sdk/android/api/org/webrtc/NativeLibraryLoader.java:23
↓ 49 callersMethodAddSample
common_audio/smoothing_filter.cc:42
↓ 49 callersMethodCreateSimulationNode
test/scenario/scenario.cc:166
↓ 49 callersFunctionHasAttribute
pc/webrtc_sdp.cc:584
↓ 49 callersMethodInsert
modules/audio_processing/aec3/echo_canceller3.cc:263
↓ 49 callersMethodMonitor
rtc_base/test_utils.h:47
↓ 49 callersMethodStart
video/video_send_stream.cc:143
↓ 49 callersMethodfake_ice_transport
Get inner fake ICE transport.
p2p/base/fake_dtls_transport.h:69
↓ 49 callersMethodset_type
p2p/base/port.h:367
↓ 48 callersFunctionDecodeDeltas
logging/rtc_event_log/encoder/delta_encoding.cc:942
↓ 48 callersFunctionEncodeDeltas
logging/rtc_event_log/encoder/delta_encoding.cc:936
↓ 48 callersMethodGetAddress
api/transport/stun.cc:143
↓ 48 callersMethodGetContentDescriptionByName
pc/session_description.cc:110
↓ 48 callersMethodInitialize
pc/peer_connection.cc:1129
↓ 48 callersFunctionMinSample
system_wrappers/source/metrics.cc:316
↓ 48 callersMethodOnReceivedPacket
modules/video_coding/nack_module.cc:109
↓ 48 callersMethodPrepareAddress
p2p/base/tcp_port.cc:166
↓ 48 callersFunctionRelease
api/stats_types.h:299
↓ 48 callersMethodSetSize
modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc:33
↓ 48 callersFunctionSize
modules/include/module_common_types.h:39
↓ 48 callersMethodchannels
Returns a pointer array to the full-band channels (or lower band channels). Usage: channels()[channel][sample]. Where: 0 <= channel < |num_allocated_c
common_audio/channel_buffer.h:67
↓ 48 callersMethodlast_message
pc/test/mock_peer_connection_observers.h:338
↓ 48 callersMethodlast_stun_buf
The last StunMessage that was sent on this Port. TODO(?): Make these const; requires changes to SendXXXXResponse.
p2p/base/port_unittest.cc:158
↓ 48 callersMethodremove
api/jsep_ice_candidate.cc:64
↓ 47 callersFunctionAddLine
pc/webrtc_sdp.cc:466
↓ 47 callersMethodAddReportBlock
modules/rtp_rtcp/source/rtcp_packet/sender_report.cc:121
↓ 47 callersMethodContinuousFrame
modules/video_coding/decoding_state.cc:200
↓ 47 callersFunctionCreateNetworkEmulationManager
api/test/create_network_emulation_manager.cc:20
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