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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 15 callersMethodexpectIceConnectionChange
(IceConnectionState newState)
sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java:175
↓ 15 callersMethodexpectStandardizedIceConnectionChange
(IceConnectionState newState)
sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java:181
↓ 15 callersMethodfirst_seq_num
modules/video_coding/frame_object.cc:98
↓ 15 callersMethodfraction_lost
modules/rtp_rtcp/source/rtcp_packet/report_block.h:51
↓ 15 callersMethodgetSrtp
()
sdk/android/api/org/webrtc/CryptoOptions.java:101
↓ 15 callersMethodgetTimestampNs
Timestamp of the frame in nano seconds.
sdk/android/api/org/webrtc/VideoFrame.java:161
↓ 15 callersMethodheight
api/video/i010_buffer.cc:165
↓ 15 callersMethodlength
modules/rtp_rtcp/source/ulpfec_generator.cc:105
↓ 15 callersMethodlogAndToast
(String msg)
examples/androidapp/src/org/appspot/apprtc/CallActivity.java:680
↓ 15 callersMethodrecv_codecs
media/engine/webrtc_video_engine.cc:483
↓ 15 callersMethodreportError
(final String errorMessage)
examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java:358
↓ 15 callersMethodrtp_timestamp
modules/rtp_rtcp/source/rtcp_packet/sender_report.h:54
↓ 15 callersMethodrtp_timestamp
api/transport/rtp/rtp_source.h:84
↓ 15 callersMethodsecond_factory
api/test/loopback_media_transport.h:97
↓ 15 callersMethodsend_bandwidth
test/scenario/call_client.h:106
↓ 15 callersMethodset_msid_signaling
Determines how the MSIDs were/will be signaled. Flag value composed of MsidSignaling bits (see enum above).
pc/session_description.h:605
↓ 15 callersMethodsignal_count
rtc_base/sigslot_unittest.cc:58
↓ 15 callersMethodsmoothed_rtt
Returns the EWMA smoothed RTT for the connection. May return Zero if no valid updates have occurred.
modules/congestion_controller/bbr/rtt_stats.h:43
↓ 15 callersMethodticks
api/neteq/tick_timer.h:88
↓ 15 callersMethodtype
api/transport/stun.h:151
↓ 15 callersMethodwaitForFramesRendered
(int timeoutMs)
examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java:128
↓ 15 callersMethodwidth
api/video/i010_buffer.cc:161
↓ 15 callersMethodwrap
Wraps existing ByteBuffers into JavaI420Buffer object without copying the contents.
sdk/android/api/org/webrtc/JavaI420Buffer.java:52
↓ 14 callersMethodAddCandidateAddress
Exposed for testing candidate building.
p2p/base/port_unittest.cc:186
↓ 14 callersMethodAddCodec
pc/session_description.h:314
↓ 14 callersMethodAddConsumer
call/adaptation/resource_adaptation_processor.h:90
↓ 14 callersMethodAddEncodedSink
pc/video_rtp_track_source.cc:51
↓ 14 callersMethodAddInt64
api/stats_types.cc:750
↓ 14 callersMethodAnalyzeRenderAudio
modules/audio_processing/residual_echo_detector.cc:59
↓ 14 callersFunctionAppendFieldTrials
Takes the current active field trials set, and appends some new trials.
video/full_stack_tests.cc:74
↓ 14 callersMethodCanInsertDtmf
media/engine/webrtc_voice_engine.cc:1990
↓ 14 callersMethodChainOperation
rtc_base/operations_chain.h:135
↓ 14 callersFunctionChannelOrder
api/audio/channel_layout.cc:203
↓ 14 callersFunctionCheckControllersOrder
|expected_order| contains the expected indices of all controllers in the vector of controllers returned by GetSortedControllers(). A negative index me
modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc:89
↓ 14 callersMethodComputeRateForInterval
rtc_base/rate_tracker.cc:36
↓ 14 callersMethodConsumeWriteBuffer
rtc_base/memory/fifo_buffer.cc:182
↓ 14 callersMethodCreate
modules/audio_coding/neteq/expand.cc:863
↓ 14 callersMethodCreate
modules/audio_processing/audio_processing_unittest.cc:2404
↓ 14 callersMethodCreateEmulatedNetworkManagerInterface
test/network/network_emulation_manager.cc:270
↓ 14 callersFunctionCreateSpecificSimulcastTestFixture
modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc:24
↓ 14 callersMethodCurrentThread
rtc_base/thread.cc:235
↓ 14 callersFunctionDbToRatio
common_audio/include/audio_util.h:73
↓ 14 callersMethodDeRegisterModule
modules/utility/source/process_thread_impl.cc:152
↓ 14 callersMethodDeliverRecordedData
modules/audio_device/fine_audio_buffer.cc:108
↓ 14 callersFunctionDesktopRect
modules/desktop_capture/desktop_geometry.h:102
↓ 14 callersMethodDestroy
p2p/base/port.cc:836
↓ 14 callersFunctionDirName
test/testsupport/file_utils.cc:70
↓ 14 callersFunctionDirectionToString
modules/audio_device/win/core_audio_base_win.cc:48
↓ 14 callersFunctionEmptySocketAddressWithFamily
rtc_base/socket_address.cc:325
↓ 14 callersMethodEnable
pc/channel.cc:264
↓ 14 callersMethodExpectAllMembersSuccessfullyTested
pc/rtc_stats_integrationtest.cc:277
↓ 14 callersFunctionFetchAllPackets
modules/rtp_rtcp/source/rtp_format_h264_unittest.cc:102
↓ 14 callersFunctionFillEncoderConfiguration
test/encoder_settings.cc:132
↓ 14 callersMethodFillTimingInfo
video/frame_encode_metadata_writer.cc:129
↓ 14 callersFunctionFindValueOrNull
pc/track_media_info_map.cc:21
↓ 14 callersMethodFlush
modules/audio_coding/neteq/dtmf_buffer.cc:31
↓ 14 callersMethodFrameDependenciesDiffs
modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc:81
↓ 14 callersMethodFrameSent
video/overuse_frame_detector.cc:620
↓ 14 callersMethodGet
modules/desktop_capture/win/scoped_gdi_object.h:30
↓ 14 callersMethodGetAdjustedBitrateBps
common_video/bitrate_adjuster.cc:65
↓ 14 callersMethodGetFreshStatsReport
pc/rtc_stats_collector_unittest.cc:388
↓ 14 callersFunctionGetIntFromJson
rtc_base/strings/json.cc:40
↓ 14 callersMethodGetNewStopwatch
Returns a new Stopwatch object, based on the current TickTimer. Note that the new Stopwatch object contains a reference to the current TickTimer, and
api/neteq/tick_timer.h:95
↓ 14 callersMethodGetPayloadPaddingPacket
modules/rtp_rtcp/source/rtp_packet_history.cc:299
↓ 14 callersFunctionGetQp
modules/video_coding/utility/vp8_header_parser.cc:164
↓ 14 callersMethodGetReceiveStreamDataCounters
modules/rtp_rtcp/source/receive_statistics_impl.cc:258
↓ 14 callersMethodGetStatistics
modules/audio_processing/audio_processing_impl.cc:2127
↓ 14 callersMethodGetStats
pc/jsep_transport.cc:400
↓ 14 callersMethodGetStats
p2p/base/p2p_transport_channel.cc:1470
↓ 14 callersFunctionGetSupportedGoogPingVersion
Utility function for testing goog ping.
p2p/base/port_unittest.cc:2707
↓ 14 callersMethodHasContentName
pc/session_description.cc:71
↓ 14 callersMethodInit
pc/data_channel.cc:177
↓ 14 callersFunctionInitializeAndroidObjects
pc/test/android_test_initializer.cc:55
↓ 14 callersMethodInsertPacket
modules/audio_coding/neteq/neteq_impl.cc:174
↓ 14 callersFunctionInterfaceAddress
rtc_base/ip_address.h:131
↓ 14 callersMethodIsIceGatheringDone
pc/peer_connection_wrapper.cc:316
↓ 14 callersMethodIsWritable
pc/rtp_transport.cc:122
↓ 14 callersFunctionMatchTypeNameWithIndexPattern
Test if the network name matches the type<number> pattern, e.g. eth0. The matching is case-sensitive.
rtc_base/network.cc:193
↓ 14 callersMethodMaxFps
rtc_base/experiments/balanced_degradation_settings.cc:418
↓ 14 callersMethodMean
rtc_base/numerics/sample_stats.cc:20
↓ 14 callersMethodMinDirectPathFilterDelay
Returns the delay estimate based on the linear filter.
modules/audio_processing/aec3/aec_state.h:94
↓ 14 callersMethodOffsetIndex
modules/audio_processing/aec3/fft_buffer.h:39
↓ 14 callersMethodOnPacketReceived
media/engine/webrtc_voice_engine.cc:2042
↓ 14 callersMethodPeekNextPacket
modules/audio_coding/neteq/packet_buffer.cc:208
↓ 14 callersMethodPerform
modules/audio_coding/test/iSACTest.cc:194
↓ 14 callersMethodProcess
Process any pending tasks such as timeouts (non time critical events).
modules/rtp_rtcp/source/rtp_rtcp_impl.cc:98
↓ 14 callersMethodPrune
p2p/base/port.cc:759
↓ 14 callersMethodRead
p2p/base/mdns_message.cc:112
↓ 14 callersMethodReadFecHeader
modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc:58
↓ 14 callersMethodReceivePacket
modules/rtp_rtcp/source/rtcp_transceiver.cc:80
↓ 14 callersMethodRegisterEncodeCompleteCallback
media/engine/simulcast_encoder_adapter_unittest.cc:220
↓ 14 callersMethodRemove
modules/audio_coding/neteq/decoder_database.cc:170
↓ 14 callersFunctionRemoveFile
test/testsupport/file_utils.cc:216
↓ 14 callersMethodRestartIce
pc/peer_connection.cc:2175
↓ 14 callersMethodRewind
modules/audio_coding/test/PCMFile.cc:212
↓ 14 callersFunctionSaturatedUsToCompactNtp
modules/rtp_rtcp/source/time_util.cc:63
↓ 14 callersMethodSaveWav
Saves wav file. Args: output_filepath: path to the wav audio track file to save. signal: AudioSegment instance.
modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py:72
↓ 14 callersFunctionSdpContentsAll
pc/sdp_utils.cc:42
↓ 14 callersMethodSeek
rtc_base/bit_buffer.cc:238
↓ 14 callersMethodSend
examples/peerconnection/server/data_socket.cc:118
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