MCPcopy Create free account

hub / github.com/HackWebRTC/webrtc / functions

Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 20 callersFunctionAddRtxCodec
pc/media_session_unittest.cc:285
↓ 20 callersMethodAdvanceTime
call/rtp_video_sender_unittest.cc:166
↓ 20 callersMethodBwHigh
rtc_base/experiments/quality_rampup_experiment.cc:55
↓ 20 callersMethodCanAdaptUp
rtc_base/experiments/balanced_degradation_settings.cc:431
↓ 20 callersMethodConstructMediaPackets
modules/rtp_rtcp/source/fec_test_helper.cc:45
↓ 20 callersFunctionContainsSender
Check if |senders| contains the specified sender, by id.
pc/peer_connection_interface_unittest.cc:519
↓ 20 callersMethodCreateLocalVideoTrack
pc/peer_connection_integrationtest.cc:340
↓ 20 callersFunctionCreateRtcpPacket
Produces a packet buffer representing an RTCP packet with a given SSRC, as it would look when sent over the wire. |distinguishing_string| allows diffe
call/rtcp_demuxer_unittest.cc:89
↓ 20 callersFunctionCreateSquareFrameGenerator
api/test/create_frame_generator.cc:23
↓ 20 callersMethodCreateTaskQueue
test/time_controller/external_time_controller.cc:200
↓ 20 callersFunctionFindNthReportByType
Finds the |n|-th report of type |type| in |reports|. |n| starts from 1 for finding the first report.
pc/stats_collector_unittest.cc:203
↓ 20 callersMethodGetBaseMinimumPlayoutDelayMs
media/engine/webrtc_voice_engine.cc:1975
↓ 20 callersMethodGetEncoderInfo
media/engine/simulcast_encoder_adapter_unittest.cc:232
↓ 20 callersFunctionGetFirstMediaContent
pc/media_session.cc:2753
↓ 20 callersFunctionGetFirstRtpDataContentDescription
pc/media_session.cc:2815
↓ 20 callersMethodGetPercentileValue
rtc_base/numerics/percentile_filter.h:112
↓ 20 callersMethodGetRtpSenderCapabilities
pc/peer_connection_factory.cc:157
↓ 20 callersMethodGetRtpTransport
pc/jsep_transport_controller.cc:134
↓ 20 callersMethodHasFeedbackParam
media/base/codec.cc:183
↓ 20 callersMethodIsAtEnd
modules/desktop_capture/desktop_region.cc:514
↓ 20 callersMethodLatestEstimate
modules/remote_bitrate_estimator/aimd_rate_control.cc:179
↓ 20 callersFunctionLatestSequenceNumber
modules/include/module_common_types_public.h:99
↓ 20 callersMethodOnEncodedImage
call/rtp_video_sender.cc:460
↓ 20 callersMethodOnReadPacket
p2p/base/port.cc:375
↓ 20 callersMethodOnRtcpPacket
call/rtcp_demuxer.cc:73
↓ 20 callersFunctionPrintCodecMappingEntry
modules/audio_coding/neteq/tools/neteq_rtpplay.cc:207
↓ 20 callersMethodRegisterObserver
pc/dtmf_sender.cc:92
↓ 20 callersMethodRelease
modules/video_coding/media_opt_util.cc:700
↓ 20 callersFunctionResetFrame
Utility function that resets the frame member variables with sensible defaults.
modules/audio_mixer/audio_mixer_impl_unittest.cc:42
↓ 20 callersFunctionRtpPacketInfo
api/rtp_packet_info.h:51
↓ 20 callersFunctionRunTest
modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc:152
↓ 20 callersMethodSet
modules/audio_device/win/core_audio_utility_win.h:295
↓ 20 callersMethodSet
modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc:119
↓ 20 callersMethodSetPort
rtc_base/socket_address.cc:123
↓ 20 callersMethodSetRecordingChannels
modules/audio_device/audio_device_buffer.cc:210
↓ 20 callersMethodStart
modules/audio_device/android/aaudio_wrapper.cc:176
↓ 20 callersFunctionTime
Deprecated. Do not use this in any new code.
rtc_base/time_utils.h:77
↓ 20 callersFunctionToString
rtc_base/strings/audio_format_to_string.cc:18
↓ 20 callersMethodTryUseRate
Usage note: This class is intended be usable in a scenario where different threads may call each of the the different method. For instance, a network
rtc_base/rate_limiter.cc:33
↓ 20 callersFunctionUpdate
Update filter parameters based on the pitch-gains and pitch-lags. */
modules/audio_coding/codecs/isac/main/source/pitch_filter.c:215
↓ 20 callersFunctionWebRtcSpl_ZerosArrayW16
common_audio/signal_processing/copy_set_operations.c:74
↓ 20 callersMethodbuild
()
sdk/android/tests/src/org/webrtc/AndroidVideoDecoderTest.java:159
↓ 20 callersMethodcapture_time_ms
Time in local time base as close as it can to frame capture time.
modules/rtp_rtcp/source/rtp_packet_to_send.h:47
↓ 20 callersMethodcurrent_frame
modules/desktop_capture/screen_capture_frame_queue.h:62
↓ 20 callersMethodgetDataU
Returns a direct ByteBuffer containing U-plane data. The buffer capacity is at least getStrideU() ((getHeight() + 1) / 2) bytes. The position of the r
sdk/android/api/org/webrtc/VideoFrame.java:79
↓ 20 callersMethodgetDataV
Returns a direct ByteBuffer containing V-plane data. The buffer capacity is at least getStrideV() ((getHeight() + 1) / 2) bytes. The position of the r
sdk/android/api/org/webrtc/VideoFrame.java:86
↓ 20 callersMethodis_open
Returns true if a file has been opened. If the file is not open, no methods but is_open and Close may be called.
rtc_base/system/file_wrapper.h:62
↓ 20 callersMethodlast_sent_packet
p2p/base/fake_ice_transport.h:265
↓ 20 callersMethodnext_index
Accessors and mutators.
modules/audio_coding/neteq/sync_buffer.h:94
↓ 20 callersMethodrelease
()
sdk/android/api/org/webrtc/VideoFrame.java:185
↓ 20 callersMethodresult
pc/test/mock_peer_connection_observers.h:292
↓ 20 callersMethodretain
()
sdk/android/api/org/webrtc/VideoFrame.java:50
↓ 20 callersMethodset_adaptation_enabled
video/video_stream_encoder_unittest.cc:275
↓ 20 callersMethodset_allow_retransmission
modules/rtp_rtcp/source/rtp_packet_to_send.h:64
↓ 20 callersFunctiontokenize_first
rtc_base/string_encode.cc:235
↓ 19 callersMethodAddFeedbackParam
media/base/codec.cc:179
↓ 19 callersMethodAddOrUpdateSink
media/base/video_broadcaster.cc:27
↓ 19 callersMethodAdvanceTime
rtc_base/rate_tracker_unittest.cc:24
↓ 19 callersMethodAnalyzeCaptureAudio
modules/audio_processing/gain_control_impl.cc:149
↓ 19 callersFunctionAttachCurrentThreadIfNeeded
sdk/android/native_api/jni/jvm.cc:17
↓ 19 callersMethodAttachmentId
Returns an ID that changes every time SetTrack() is called, but otherwise remains constant. Used to generate IDs for stats. The special value zero mea
pc/rtp_sender.h:146
↓ 19 callersMethodCanAdaptUpResolution
rtc_base/experiments/balanced_degradation_settings.cc:441
↓ 19 callersMethodClear
p2p/base/stun_request.cc:107
↓ 19 callersMethodCopyTo
modules/audio_coding/neteq/audio_vector.cc:40
↓ 19 callersMethodCreateEmulatedNode
test/network/network_emulation_manager.cc:60
↓ 19 callersMethodCreateEndpoint
test/network/network_emulation_manager.cc:81
↓ 19 callersFunctionCreateIceCandidate
pc/jsep_ice_candidate.cc:19
↓ 19 callersMethodCreateProbeCluster
modules/pacing/paced_sender.cc:58
↓ 19 callersFunctionCreateRandomUuid
Version 4 UUID is of the form: xxxxxxxx-xxxx-4xxx-yxxx-xxxxxxxxxxxx Where 'x' is a hex digit, and 'y' is 8, 9, a or b.
rtc_base/helpers.cc:165
↓ 19 callersMethodFindNextConfiguration
call/adaptation/resource_adaptation_processor.cc:50
↓ 19 callersMethodFutureLength
modules/audio_coding/neteq/sync_buffer.cc:19
↓ 19 callersMethodGetClock
api/test/create_time_controller_unittest.cc:42
↓ 19 callersFunctionIPIsLinkLocal
rtc_base/ip_address.cc:480
↓ 19 callersMethodIncrement
api/neteq/tick_timer.h:83
↓ 19 callersMethodInit
modules/audio_coding/neteq/post_decode_vad.cc:39
↓ 19 callersMethodIsComfortNoise
modules/audio_coding/neteq/decoder_database.cc:276
↓ 19 callersMethodLoadWav
Loads wav file. Args: filepath: path to the wav audio track file to load. channels: number of channels (downmixing to mono by default
modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py:55
↓ 19 callersMethodMixSignals
Mixes |signal| and |noise| with a target SNR. Mix |signal| and |noise| with a desired SNR by scaling |noise|. If the target SNR is +/- infini
modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py:293
↓ 19 callersMethodNormalized
rtc_base/ip_address.cc:170
↓ 19 callersMethodParseArgs
(self, argv)
tools_webrtc/mb/mb.py:77
↓ 19 callersFunctionParseError
pc/sdp_serializer.cc:53
↓ 19 callersMethodProcess
modules/audio_coding/neteq/merge.cc:46
↓ 19 callersMethodRand<float>
rtc_base/random.cc:50
↓ 19 callersMethodRegisterModule
modules/utility/source/process_thread_impl.cc:116
↓ 19 callersMethodRegisterPayload
modules/audio_coding/neteq/decoder_database.cc:155
↓ 19 callersMethodRenderTimeMs
modules/video_coding/timing.cc:172
↓ 19 callersFunctionSdpDeserializeCandidate
pc/webrtc_sdp_unittest.cc:923
↓ 19 callersMethodSendData
media/base/rtp_data_engine.cc:256
↓ 19 callersFunctionSetBE32
rtc_base/byte_order.h:113
↓ 19 callersMethodSetCandidateFilter
p2p/client/basic_port_allocator.cc:304
↓ 19 callersMethodSetOfferAnswerOptions
Sets the options to be used when CreateAndSetAndSignalOffer is called, or when a remote offer is received (via fake signaling) and an answer is genera
pc/peer_connection_integrationtest.cc:244
↓ 19 callersMethodSetPlayoutChannels
modules/audio_device/audio_device_buffer.cc:216
↓ 19 callersMethodSetSendingStatus
modules/rtp_rtcp/source/rtcp_sender.cc:228
↓ 19 callersMethodState
modules/congestion_controller/goog_cc/trendline_estimator.cc:266
↓ 19 callersFunctionToStringIfSet
api/audio_options.cc:20
↓ 19 callersFunctionUpdateRateControl
modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc:55
↓ 19 callersMethodUpdateRtt
Updates the RTT based on a new sample.
modules/congestion_controller/bbr/rtt_stats.cc:47
↓ 19 callersMethodWaitUntilTaskQueueIsIdle
This is used as a synchronisation mechanism, to make sure that the encoder queue is not blocked before we start sending it frames.
video/video_stream_encoder_unittest.cc:179
↓ 19 callersMethodWasDecoded
modules/video_coding/utility/decoded_frames_history.cc:73
↓ 19 callersFunctionWebRtcSpl_NormW16
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:141
← previousnext →1,101–1,200 of 33,939, ranked by callers