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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 129 callersMethodLength
Returns number of unprocessed bytes.
rtc_base/byte_buffer.h:135
↓ 126 callersMethodGetLocalAddress
rtc_base/async_socket.cc:41
↓ 126 callersMethodRand
rtc_base/random.cc:24
↓ 126 callersMethodtimestamp
pc/test/mock_peer_connection_observers.h:391
↓ 125 callersMethodReadBits
rtc_base/bit_buffer.cc:144
↓ 125 callersMethodcapacity
rtc_base/buffer.h:148
↓ 125 callersMethodmid
pc/rtp_transceiver.cc:233
↓ 124 callersMethodBuild
modules/rtp_rtcp/source/rtcp_packet.cc:20
↓ 123 callersMethodError
rtc_base/socket_adapters.cc:457
↓ 123 callersMethodbegin
api/stats_types.cc:793
↓ 121 callersMethodTestMemberIsUndefined
pc/rtc_stats_integrationtest.cc:228
↓ 120 callersMethodcdata
api/test/network_emulation/network_emulation_interfaces.h:39
↓ 119 callersMethodtimestamp_us
api/rtc_event_log/rtc_event.h:66
↓ 118 callersMethodInsertPacket
modules/video_coding/receiver.cc:56
↓ 117 callersMethodSetSequenceNumber
modules/rtp_rtcp/source/rtp_sender.cc:582
↓ 116 callersMethodAddAudioVideoTracks
pc/peer_connection_integrationtest.cc:315
↓ 116 callersFunctionInvoke
rtc_base/thread.h:317
↓ 115 callersMethoddata_channel
pc/peer_connection_integrationtest.cc:406
↓ 114 callersFunctionInsert
modules/audio_processing/vad/vad_circular_buffer_unittest.cc:36
↓ 114 callersMethodReadExponentialGolomb
rtc_base/bit_buffer.cc:185
↓ 114 callersMethodobj
sdk/android/native_api/jni/scoped_java_ref.h:60
↓ 113 callersFunctionBind
rtc_base/bind.h:220
↓ 113 callersMethodSequenceNumber
modules/rtp_rtcp/source/rtp_sender.cc:600
↓ 113 callersMethoddata_observer
pc/peer_connection_integrationtest.cc:407
↓ 112 callersMethodIsActive
pc/srtp_filter.cc:28
↓ 112 callersMethodget
api/scoped_refptr.h:105
↓ 111 callersMethodDumpRaw
Methods for performing dumping of data of various types into various formats.
modules/audio_processing/logging/apm_data_dumper.h:82
↓ 111 callersMethodSetVideoSend
media/engine/webrtc_video_engine.cc:1089
↓ 111 callersMethodbegin
modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc:118
↓ 111 callersMethodtimestamp_ms
api/rtc_event_log/rtc_event.h:65
↓ 110 callersMethodSize
Returns the number of elements in this AudioVector.
modules/audio_coding/neteq/audio_vector.cc:276
↓ 110 callersMethodconn
p2p/base/port_unittest.cc:275
↓ 109 callersMethodParse
test/rtp_header_parser.cc:63
↓ 109 callersMethodSetTimestamp
modules/rtp_rtcp/source/rtp_packet.cc:151
↓ 108 callersMethodAddTransceiver
pc/peer_connection.cc:1699
↓ 107 callersMethoderror
pc/data_channel.cc:264
↓ 106 callersMethodEncode
modules/video_coding/codecs/vp9/vp9_impl.cc:835
↓ 106 callersMethodGetFrame
media/base/fake_frame_source.cc:61
↓ 104 callersMethodCheck
rtc_base/test_utils.h:56
↓ 104 callersMethodGetContentByName
pc/session_description.cc:101
↓ 103 callersMethoddata
rtc_base/buffer.h:125
↓ 103 callersMethodpc
pc/peer_connection_integrationtest.cc:230
↓ 103 callersMethodus
api/units/timestamp.h:57
↓ 102 callersFunctionSendTask
rtc_base/task_queue_for_test.h:28
↓ 102 callersMethodVP8
api/video_codecs/video_codec.cc:102
↓ 101 callersFunctionCheckedDivExact
rtc_base/checks.h:442
↓ 101 callersMethodDetach
rtc_base/stream.cc:115
↓ 101 callersFunctionSdpDeserialize
pc/webrtc_sdp_unittest.cc:918
↓ 99 callersMethodOnRtpPacket
call/rtp_demuxer.cc:174
↓ 99 callersMethodSet
rtc_base/event.cc:46
↓ 99 callersMethodmutable_data
TODO(henrik.lundin) Can we skip zeroing the buffer? See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
api/audio/audio_frame.cc:116
↓ 99 callersMethodstate
pc/remote_audio_source.cc:92
↓ 99 callersMethodwidth
modules/desktop_capture/desktop_geometry.h:57
↓ 98 callersMethodWait
rtc_base/event.cc:54
↓ 97 callersMethodAddAttribute
api/transport/stun.cc:97
↓ 97 callersFunctionOutputPath
test/testsupport/file_utils.cc:94
↓ 97 callersFunctionTimeMicros
rtc_base/time_utils.cc:218
↓ 97 callersFunctionWebRtcSpl_NormW32
Return the number of steps a can be left-shifted without overflow, or 0 if a == 0.
common_audio/signal_processing/include/spl_inl.h:129
↓ 96 callersMethodOnBitrateUpdated
video/video_stream_encoder_unittest.cc:2846
↓ 96 callersMethodOnRtpPacket
modules/rtp_rtcp/source/flexfec_receiver.cc:60
↓ 95 callersMethodParse
modules/rtp_rtcp/source/rtp_packet.cc:77
↓ 95 callersMethodat
api/jsep_ice_candidate.cc:50
↓ 95 callersMethodprotocol
pc/data_channel.h:142
↓ 94 callersMethodAdvanceTimeMilliseconds
If |stop_on_frame| is true and next frame arrives between now and now+|milliseconds|, the clock will be advanced to the arrival time of next frame. Ot
modules/video_coding/receiver_unittest.cc:260
↓ 93 callersMethodAddSample
rtc_base/numerics/sample_stats.cc:56
↓ 93 callersMethodheight
modules/desktop_capture/desktop_geometry.h:58
↓ 92 callersMethodlog_time_us
logging/rtc_event_log/logged_events.h:263
↓ 92 callersMethodnum_packets
Returns total number of rtcp packet received.
modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc:108
↓ 91 callersMethoddirection
pc/rtp_transceiver.cc:277
↓ 91 callersMethodtrack
pc/rtp_sender.h:90
↓ 91 callersMethodvideo_frame_buffer
api/video/video_frame.cc:301
↓ 90 callersMethodSendTask
test/scenario/call_client.cc:317
↓ 89 callersMethodClose
modules/audio_coding/test/RTPFile.cc:142
↓ 89 callersFunctionNumBandsForRate
TODO(peah): Integrate this with how it is done inside audio_processing_impl.
modules/audio_processing/aec3/aec3_common.h:56
↓ 89 callersMethodOnAdaptationChanged
video/send_statistics_proxy.cc:1055
↓ 89 callersMethodTimestamp
modules/rtp_rtcp/source/rtp_packet.h:58
↓ 88 callersFunctionAddMediaDescriptionOptions
Add a media section to the |session_options|.
pc/media_session_unittest.cc:321
↓ 88 callersMethodGet
rtc_base/experiments/field_trial_parser.cc:179
↓ 88 callersMethodIsFinite
rtc_base/units/unit_base.h:44
↓ 88 callersMethodSetRecvParameters
media/engine/webrtc_voice_engine.cc:1325
↓ 88 callersMethodmedia_type
pc/rtp_transceiver.cc:229
↓ 87 callersMethodGetRtpSendParameters
media/engine/webrtc_voice_engine.cc:1352
↓ 87 callersMethodset_video_frame_buffer
api/video/video_frame.cc:305
↓ 86 callersMethodIsNil
rtc_base/ip_address.cc:56
↓ 86 callersMethodsender_ssrc
modules/rtp_rtcp/source/rtcp_packet.h:62
↓ 85 callersMethodCreateAsyncSocket
rtc_base/null_socket_server.cc:37
↓ 84 callersFunctionReadParam
api/audio/echo_canceller3_config_json.cc:24
↓ 84 callersMethodToString
pc/jsep_ice_candidate.cc:56
↓ 84 callersMethodWait
modules/video_coding/receiver_unittest.cc:353
↓ 83 callersFunctionIsEnabled
modules/pacing/pacing_controller.cc:48
↓ 83 callersMethodSetRtpSendParameters
media/engine/webrtc_voice_engine.cc:1372
↓ 83 callersMethodToString
rtc_base/network.cc:1070
↓ 82 callersMethodGetAudio
modules/audio_coding/neteq/neteq_impl.cc:241
↓ 82 callersFunctionParseFieldTrial
rtc_base/experiments/field_trial_parser.cc:40
↓ 82 callersMethodat
pc/stream_collection.h:39
↓ 82 callersMethodbuild
api/video/video_frame.cc:164
↓ 80 callersMethodAdaptFrameResolution
media/base/video_adapter.cc:188
↓ 80 callersMethodAddTrack
pc/media_stream.cc:41
↓ 80 callersFunctionGetFirstVideoContentDescription
pc/media_session.cc:2809
↓ 80 callersFunctionLimit
api/audio/echo_canceller3_config.cc:20
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