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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 31 callersMethodGetWeakPtr
rtc_base/weak_ptr.h:246
↓ 31 callersFunctionIPIsPrivate
rtc_base/ip_address.cc:333
↓ 31 callersMethodParse
modules/rtp_rtcp/source/rtp_packet_unittest.cc:767
↓ 31 callersMethodPostDelayedTask
rtc_base/thread.cc:962
↓ 31 callersFunctionRandomizeSampleVector
modules/audio_processing/test/echo_canceller_test_tools.cc:17
↓ 31 callersMethodReset
modules/audio_processing/aec3/aec_state.cc:297
↓ 31 callersMethodRtcpSent
modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc:124
↓ 31 callersFunctionRtpTransceiverDirectionHasRecv
pc/rtp_media_utils.cc:35
↓ 31 callersMethodSetEstimatedBitrate
modules/congestion_controller/goog_cc/alr_detector.cc:100
↓ 31 callersMethodSetSuggestedYAxis
rtc_tools/rtc_event_log_visualizer/plot_base.cc:55
↓ 31 callersMethodSpectrum
Computes the power spectrum of the data.
modules/audio_processing/aec3/fft_data.h:44
↓ 31 callersMethodToJson
stats/rtc_stats.cc:100
↓ 31 callersMethodTriggerCpuOveruse
video/video_stream_encoder_unittest.cc:185
↓ 31 callersFunctionUpdateNetworkMetrics
modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc:125
↓ 31 callersMethodbitrate_bps
logging/rtc_event_log/events/rtc_event_probe_result_success.h:34
↓ 31 callersMethodcheck
(self, args, mbw=None, files=None, out=None, err=None, ret=None)
tools_webrtc/mb/mb_unittest.py:179
↓ 31 callersMethoddtls_transport
p2p/base/dtls_transport_unittest.cc:108
↓ 31 callersMethodfake_frame
modules/audio_mixer/audio_mixer_impl_unittest.cc:83
↓ 31 callersMethodgeneration
Identifies the generation that this port was created in.
p2p/base/port.h:236
↓ 31 callersMethodgetBuffer
()
sdk/android/api/org/webrtc/VideoFrame.java:145
↓ 31 callersMethodrect
modules/desktop_capture/desktop_frame.cc:106
↓ 31 callersMethodset_video_codecs
pc/media_session.cc:1375
↓ 31 callersMethody
modules/desktop_capture/desktop_geometry.h:27
↓ 30 callersMethodAddInterface
p2p/client/basic_port_allocator_unittest.cc:168
↓ 30 callersMethodAddReceivedPacket
TODO(eladalon): Consider using packet.recovered() to avoid processing recovered packets here.
modules/rtp_rtcp/source/flexfec_receiver.cc:86
↓ 30 callersMethodCreateVideoStream
test/scenario/scenario.cc:213
↓ 30 callersFunctionDecodeTest
rtc_base/base64_unittest.cc:1365
↓ 30 callersFunctionExtractStatsValue
pc/stats_collector_unittest.cc:165
↓ 30 callersFunctionFindFirstMediaDescriptionByMid
This is used for test only. MIDs are not the identification of the MediaDescriptionOptions since some end points may not support MID and the SDP may n
pc/media_session_unittest.cc:304
↓ 30 callersMethodFindMappingFor
media/engine/payload_type_mapper.cc:111
↓ 30 callersMethodGetAverage
Returns average in O(1) time. This function may not be called if there are no samples.
rtc_base/numerics/samples_stats_counter.h:63
↓ 30 callersFunctionGetCandidate
p2p/base/port_unittest.cc:107
↓ 30 callersMethodGetSenders
pc/peer_connection.cc:1972
↓ 30 callersFunctionIPIsLoopback
rtc_base/ip_address.cc:321
↓ 30 callersMethodLog
logging/rtc_event_log/fake_rtc_event_log.cc:34
↓ 30 callersMethodSetIceTiebreaker
p2p/base/port.cc:221
↓ 30 callersMethodSetParam
media/base/codec.cc:167
↓ 30 callersMethodSize
rtc_base/ip_address.cc:60
↓ 30 callersFunctionUQ32x32ToInt64Ms
Converts UQ32.32-formatted fixed-point seconds to |int64_t| milliseconds.
system_wrappers/include/ntp_time.h:110
↓ 30 callersMethodUnwrap
rtc_base/numerics/sequence_number_util.h:95
↓ 30 callersMethodcbegin
rtc_base/buffer.h:198
↓ 30 callersMethodcheckIsOnMainThread
Throws exception if called from other than main thread.
rtc_base/java/src/org/webrtc/ThreadUtils.java:45
↓ 30 callersMethodremb
test/rtcp_packet_parser.h:99
↓ 30 callersMethodrender_time_ms
api/video/video_frame.cc:311
↓ 30 callersMethodset_extmap_allow_mixed_enum
pc/session_description.h:220
↓ 30 callersMethodset_protocol
pc/session_description.h:98
↓ 30 callersMethodset_stream_ids
pc/video_rtp_receiver.cc:182
↓ 30 callersMethodsocket
p2p/base/turn_port.h:201
↓ 29 callersMethodAccept
rtc_base/async_socket.cc:82
↓ 29 callersMethodAppend
modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc:22
↓ 29 callersMethodCreateConnection
p2p/base/tcp_port.cc:119
↓ 29 callersFunctionCreateLayers
pc/peer_connection_simulcast_unittest.cc:65
↓ 29 callersFunctionDivideRoundToNearest
rtc_base/numerics/divide_round.h:34
↓ 29 callersMethodEncode
modules/audio_coding/codecs/cng/webrtc_cng.cc:236
↓ 29 callersMethodGetLifetimeStatistics
modules/audio_coding/neteq/neteq_impl.cc:361
↓ 29 callersMethodGetParam
media/base/codec.cc:152
↓ 29 callersMethodGetType
TODO(steveanton): Remove this default implementation once Chromium has been updated.
pc/jsep_session_description.cc:120
↓ 29 callersMethodGetUInt32
api/transport/stun.cc:160
↓ 29 callersMethodLength
modules/audio_coding/neteq/dtmf_buffer.cc:187
↓ 29 callersMethodOnPacketsFeedback
modules/congestion_controller/pcc/rtt_tracker.cc:21
↓ 29 callersMethodPrepareCaptureProcessing
Prepares the render buffers for processing another capture block.
modules/audio_processing/aec3/render_delay_buffer.cc:247
↓ 29 callersMethodProcessMessagesUntilIdle
rtc_base/virtual_socket_server.cc:633
↓ 29 callersFunctionRotateDesktopFrame
modules/desktop_capture/desktop_frame_rotation.cc:93
↓ 29 callersMethodRtcpReportBlocks
modules/rtp_rtcp/source/receive_statistics_impl.cc:365
↓ 29 callersFunctionSafeSetError
pc/channel.cc:91
↓ 29 callersMethodSendTo
rtc_base/test_client.cc:56
↓ 29 callersMethodSetBase
modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc:297
↓ 29 callersMethodSetData
modules/rtp_rtcp/source/rtcp_packet/app.cc:68
↓ 29 callersMethodSetDestination
Simulates the two transports connecting to each other. If |asymmetric| is true this method only affects this FakeIceTransport. If false, it affects |d
p2p/base/fake_ice_transport.h:56
↓ 29 callersMethodSetPacingRates
modules/pacing/paced_sender.cc:105
↓ 29 callersMethodSetRemoteIceParameters
p2p/base/p2p_transport_channel.cc:424
↓ 29 callersMethodSetWritable
p2p/base/p2p_transport_channel.cc:2053
↓ 29 callersMethodStop
rtc_base/thread.cc:817
↓ 29 callersFunctionTimeMicrosToNtp
modules/rtp_rtcp/source/time_util.cc:37
↓ 29 callersMethodUpdate
modules/audio_processing/echo_detector/moving_max.cc:31
↓ 29 callersMethodUpdateLastReceivedPacket
modules/audio_coding/neteq/nack_tracker.cc:51
↓ 29 callersFunctionUpdateNetworkMetrics
modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc:81
↓ 29 callersFunctionWebRtcOpus_EncoderFree
modules/audio_coding/codecs/opus/opus_interface.cc:139
↓ 29 callersFunctionWebRtcSpl_GetSizeInBits
common_audio/signal_processing/include/spl_inl.h:123
↓ 29 callersMethodcreate
Create a new context with the specified config attributes, sharing data with |sharedContext|. If |sharedContext| is null, a root context is created. T
sdk/android/api/org/webrtc/EglBase.java:153
↓ 29 callersMethodgetSdp
()
sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java:628
↓ 29 callersMethodnum_packets
modules/video_coding/jitter_buffer.cc:204
↓ 29 callersMethodpacket_type
modules/rtp_rtcp/source/rtp_packet_to_send.h:52
↓ 29 callersMethodpackets_sent
modules/rtp_rtcp/source/rtp_sender_unittest.cc:124
↓ 29 callersMethodpayload
modules/rtp_rtcp/source/rtcp_packet/common_header.h:34
↓ 29 callersMethodpayload_size
Payload.
modules/rtp_rtcp/source/rtp_packet.h:65
↓ 29 callersMethodremote_description
pc/peer_connection.cc:4378
↓ 28 callersFunctionCreateCodec
modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc:54
↓ 28 callersMethodGetConfiguration
pc/peer_connection.cc:3886
↓ 28 callersMethodGetRtpVideoHeader
call/rtp_payload_params.cc:165
↓ 28 callersMethodGetStats
pc/stats_collector_unittest.cc:81
↓ 28 callersFunctionInitialize
This should be called on the NetworkManager's thread before the NetworkManager is used. Subclasses may override this if necessary.
rtc_base/network.h:113
↓ 28 callersMethodIsOk
modules/audio_coding/codecs/cng/audio_encoder_cng.cc:293
↓ 28 callersFunctionJavaToNativeString
Given a jstring, reinterprets it to a new native string.
sdk/android/native_api/jni/java_types.cc:152
↓ 28 callersMethodMarker
Header.
modules/rtp_rtcp/source/rtp_packet.h:55
↓ 28 callersMethodNetwork
p2p/base/port.cc:209
↓ 28 callersMethodOnCaptureResult
modules/desktop_capture/cropping_window_capturer.cc:80
↓ 28 callersMethodPrint
(self, *args, **kwargs)
tools_webrtc/mb/mb.py:1165
↓ 28 callersMethodReadOrDefaultValue
test/fuzzers/fuzz_data_helper.h:43
↓ 28 callersMethodSetData
rtc_base/buffer.h:206
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