MCPcopy Create free account

hub / github.com/HackWebRTC/webrtc / functions

Functions33,939 in github.com/HackWebRTC/webrtc

↓ 35 callersMethodcapacity
modules/rtp_rtcp/source/rtp_packet.h:76
↓ 35 callersFunctionend
api/rtp_packet_infos.h:69
↓ 35 callersMethodfloat_frame_view
modules/audio_processing/agc2/vector_float_frame.h:28
↓ 35 callersMethodheight
modules/video_coding/codecs/multiplex/augmented_video_frame_buffer.cc:50
↓ 35 callersMethodnum_channels
api/audio/audio_frame.h:103
↓ 35 callersMethodrbegin
logging/rtc_event_log/rtc_event_log_parser.h:184
↓ 35 callersMethodstats
p2p/base/connection.cc:1202
↓ 34 callersMethodAddAudioTrack
pc/peer_connection_integrationtest.cc:320
↓ 34 callersFunctionCheckRunOnValidThreadIfInitialized
The following thread checks are only done in DCHECK for the consistency with the exsiting thread checks.
p2p/base/port_allocator.h:622
↓ 34 callersMethodChromaWidth
api/video/video_frame_buffer.cc:42
↓ 34 callersMethodEmpty
modules/audio_coding/neteq/dtmf_buffer.cc:191
↓ 34 callersFunctionGetFirstVideoContent
pc/media_session.cc:2767
↓ 34 callersFunctionIPIsAny
rtc_base/ip_address.cc:300
↓ 34 callersMethodLocalPath
(self)
presubmit_test_mocks.py:123
↓ 34 callersMethodNextPacket
modules/rtp_rtcp/source/rtp_format_vp8.cc:76
↓ 34 callersMethodPostDelayed
rtc_base/thread.cc:511
↓ 34 callersMethodRegisterPayloadType
modules/audio_coding/neteq/neteq_impl.cc:274
↓ 34 callersMethodRelease
TODO(henrika): does not call delete this.
modules/audio_device/win/core_audio_base_win.cc:785
↓ 34 callersMethodSetLocalJsepTransportDescription
pc/jsep_transport.cc:174
↓ 34 callersMethodSetRemoteJsepTransportDescription
pc/jsep_transport.cc:269
↓ 34 callersMethodSetSize
Sets the size of the buffer. If the new size is smaller than the old, the buffer contents will be kept but truncated; if the new size is greater, the
rtc_base/buffer.h:322
↓ 34 callersMethodSetTime
rtc_base/fake_clock.cc:23
↓ 34 callersMethodStrideY
common_video/video_frame_buffer.cc:56
↓ 34 callersMethoddata
modules/rtp_rtcp/source/rtcp_packet/app.h:41
↓ 34 callersMethodsdp_mline_index
api/jsep_ice_candidate.cc:25
↓ 34 callersMethodsend_codecs
media/engine/webrtc_video_engine.cc:479
↓ 33 callersMethodAdd
modules/video_coding/histogram.cc:27
↓ 33 callersMethodAddSendRtpModule
modules/pacing/packet_router.cc:55
↓ 33 callersFunctionBuildSinkWants
media/base/video_adapter_unittest.cc:32
↓ 33 callersFunctionCheckDecision
modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc:150
↓ 33 callersFunctionGetBE16
rtc_base/byte_order.h:121
↓ 33 callersMethodIsSpatialLayerUsed
Whether the specific spatial layers has the bitrate set in any of its temporal layers.
api/video/video_bitrate_allocation.cc:61
↓ 33 callersMethodOnSendPacket
modules/rtp_rtcp/source/absolute_capture_time_sender.cc:42
↓ 33 callersFunctionReplace
pc/webrtc_sdp_unittest.cc:935
↓ 33 callersMethodSendVideo
modules/rtp_rtcp/source/rtp_sender_video.cc:493
↓ 33 callersMethodSetTitle
rtc_tools/rtc_event_log_visualizer/plot_base.cc:74
↓ 33 callersMethodSetTrack
pc/rtp_sender.cc:215
↓ 33 callersMethodStart
rtc_base/thread_unittest.cc:118
↓ 33 callersMethodType
rtc_base/checks.h:138
↓ 33 callersFunctionWebRtcSpl_NormU32
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:135
↓ 33 callersMethodWriteUInt16
rtc_base/bit_buffer.cc:255
↓ 33 callersMethodas_sctp
pc/session_description.h:85
↓ 33 callersMethodcandidates
pc/jsep_session_description.cc:283
↓ 33 callersMethodchannel
Returns the Voice/VideoChannel set for this transceiver. May be null if the transceiver is not in the currently set local/remote description.
pc/rtp_transceiver.h:76
↓ 33 callersMethodcodec_config
video/video_stream_encoder_unittest.cc:686
↓ 33 callersMethodconnections
p2p/base/p2p_transport_channel.cc:1510
↓ 33 callersMethodequals
modules/desktop_capture/desktop_geometry.h:62
↓ 33 callersMethodgetState
()
examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java:80
↓ 33 callersMethodheight
test/fake_texture_frame.cc:41
↓ 33 callersFunctionsplit
rtc_base/string_encode.cc:278
↓ 33 callersMethodtransaction_id
logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h:46
↓ 33 callersMethodtype
logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h:44
↓ 33 callersMethodwidth
modules/video_coding/codecs/multiplex/augmented_video_frame_buffer.cc:46
↓ 32 callersMethodAddString
api/stats_types.cc:736
↓ 32 callersMethodAllowRetransmission
modules/rtp_rtcp/source/rtp_sender_video_unittest.cc:130
↓ 32 callersMethodCallBeginTimeSec
rtc_tools/rtc_event_log_visualizer/analyzer.h:36
↓ 32 callersMethodCallEndTimeSec
rtc_tools/rtc_event_log_visualizer/analyzer.h:38
↓ 32 callersMethodCreateMediaChannel
media/engine/webrtc_voice_engine.cc:304
↓ 32 callersMethodCurrentTime
test/time_controller/simulated_time_controller.cc:128
↓ 32 callersMethodDecodeFec
modules/rtp_rtcp/source/forward_error_correction.cc:742
↓ 32 callersFunctionExpectParseFailure
Expect fail to parase |bad_sdp| and expect |bad_part| be part of the error message.
pc/webrtc_sdp_unittest.cc:943
↓ 32 callersMethodFindAndSetIdUsed
pc/used_ids.h:36
↓ 32 callersMethodGetDtlsTransport
pc/jsep_transport_controller.cc:172
↓ 32 callersMethodGetLocalAddress
p2p/base/turn_port.cc:307
↓ 32 callersMethodGetPeerLocalAddress
test/network/network_emulation.cc:274
↓ 32 callersMethodGetThirdBest
modules/congestion_controller/bbr/windowed_filter.h:150
↓ 32 callersFunctionGetValue
Get value only from <attribute>:<value>.
pc/webrtc_sdp.cc:615
↓ 32 callersFunctionInitFieldTrialsFromString
Optionally initialize field trial from a string.
system_wrappers/source/field_trial.cc:139
↓ 32 callersFunctionIsNewerTimestamp
NB: Doesn't fulfill strict weak ordering requirements. Mustn't be used as std::map Compare function.
modules/include/module_common_types_public.h:95
↓ 32 callersMethodMakeAudioDecoder
modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc:25
↓ 32 callersMethodOnReceivePacket
modules/rtp_rtcp/source/absolute_capture_time_receiver.cc:47
↓ 32 callersFunctionParseProfileLevelId
media/base/h264_profile_level_id.cc:122
↓ 32 callersMethodRand
modules/video_coding/frame_buffer2_unittest.cc:231
↓ 32 callersMethodRemoveSendRtpModule
modules/pacing/packet_router.cc:96
↓ 32 callersFunctionReportConditionalErrorAndExit
modules/audio_processing/test/audioproc_float_impl.cc:451
↓ 32 callersMethodSetGofInfoVP9
modules/video_coding/codecs/vp9/include/vp9_globals.h:44
↓ 32 callersMethodSetLocalCertificate
pc/jsep_transport_controller.cc:250
↓ 32 callersMethodSetMarker
modules/rtp_rtcp/source/rtp_packet.cc:131
↓ 32 callersMethodSetPayloadSize
modules/rtp_rtcp/source/rtp_packet.cc:397
↓ 32 callersMethodSetXAxis
rtc_tools/rtc_event_log_visualizer/plot_base.cc:19
↓ 32 callersMethodSsrc
audio/audio_receive_stream.cc:296
↓ 32 callersMethodStrideU
common_video/video_frame_buffer.cc:58
↓ 32 callersMethodStrideV
common_video/video_frame_buffer.cc:60
↓ 32 callersMethodToString
call/call.cc:401
↓ 32 callersMethodTriggerQualityHigh
video/video_stream_encoder_unittest.cc:206
↓ 32 callersMethodadd_ssrc
media/base/stream_params.h:120
↓ 32 callersMethodassertTrue
(boolean condition)
modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java:463
↓ 32 callersMethodpriority
p2p/base/connection.cc:328
↓ 32 callersMethodsdp_mid
api/jsep_ice_candidate.cc:21
↓ 32 callersMethodsocketserver
rtc_base/thread.cc:357
↓ 32 callersMethodsplit_bands
modules/audio_processing/audio_buffer.h:91
↓ 32 callersMethodswap
rtc_base/buffer.h:349
↓ 32 callersMethodx
modules/desktop_capture/desktop_geometry.h:26
↓ 31 callersFunctionAheadOf
rtc_base/numerics/sequence_number_util.h:61
↓ 31 callersFunctionCheckDecision
Checks that the FEC decision and |uplink_packet_loss_fraction| given by |states->controller->MakeDecision| matches |expected_enable_fec| and |expected
modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc:106
↓ 31 callersMethodChromaHeight
api/video/video_frame_buffer.cc:46
↓ 31 callersMethodCreateRoute
test/network/network_emulation.cc:151
↓ 31 callersMethodCurrentTime
test/drifting_clock.cc:31
↓ 31 callersMethodEmplace
modules/congestion_controller/bbr/packet_number_indexed_queue.h:142
↓ 31 callersMethodGetString
api/transport/stun.h:449
← previousnext →601–700 of 33,939, ranked by callers