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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 17 callersFunctionExpectNear
rtc_tools/frame_analyzer/video_color_aligner_unittest.cc:29
↓ 17 callersMethodGetArrayView
Return an array view onto the array with a given delay. A view on the last and least recently push array is returned when |delay| is 0 and N - 1 respe
modules/audio_processing/agc2/rnn_vad/ring_buffer.h:48
↓ 17 callersMethodGetBaseSequence
modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc:371
↓ 17 callersMethodGetCandidatesByMline
pc/test/mock_peer_connection_observers.h:207
↓ 17 callersMethodGetEncoderInfo
api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc:132
↓ 17 callersMethodGetFrame
rtc_tools/frame_analyzer/video_temporal_aligner.cc:127
↓ 17 callersFunctionGetLineWithType
pc/webrtc_sdp.cc:570
↓ 17 callersFunctionGetMediaConfig
media/engine/webrtc_video_engine_unittest.cc:185
↓ 17 callersMethodGetNackList
modules/video_coding/jitter_buffer.cc:618
↓ 17 callersMethodGetPacketAndSetSendTime
modules/rtp_rtcp/source/rtp_packet_history.cc:168
↓ 17 callersMethodGetSource
pc/audio_track.cc:46
↓ 17 callersMethodGetState
Get current state of frame
modules/video_coding/frame_buffer.cc:254
↓ 17 callersMethodGetTransportControllerSend
call/call.cc:1065
↓ 17 callersMethodGetVoiceChannel
media/base/fake_media_engine.cc:646
↓ 17 callersMethodIncomingPacketFeedbackVector
modules/congestion_controller/goog_cc/delay_based_bwe.cc:105
↓ 17 callersMethodIsHigh
video/quality_threshold.cc:78
↓ 17 callersMethodMakeAudioEncoder
modules/audio_coding/codecs/opus/audio_encoder_opus.cc:338
↓ 17 callersMethodMax
rtc_base/numerics/sample_stats.cc:14
↓ 17 callersMethodOnCompleteFrame
video/video_receive_stream.cc:555
↓ 17 callersMethodOnReceivedPayloadData
video/rtp_video_stream_receiver.cc:329
↓ 17 callersMethodOnRtpPacket
This method handles both regular RTP packets and packets recovered via FlexFEC.
video/rtp_video_stream_receiver.cc:551
↓ 17 callersMethodPlayoutData10Ms
Get 10 milliseconds of raw audio data to play out. Automatic resample to the requested frequency.
modules/audio_coding/acm2/audio_coding_module.cc:594
↓ 17 callersMethodProcess
modules/congestion_controller/goog_cc/probe_controller.cc:363
↓ 17 callersMethodProcess
modules/audio_processing/agc2/limiter.cc:106
↓ 17 callersMethodPushExpectedLogLine
call/bitrate_estimator_tests.cc:42
↓ 17 callersMethodReSendPacket
modules/rtp_rtcp/source/rtp_sender.cc:230
↓ 17 callersMethodRead
modules/audio_coding/test/RTPFile.cc:104
↓ 17 callersMethodRegisterTransportCallback
Register a transport callback which will be called to deliver the encoded buffers.
modules/audio_coding/acm2/audio_coding_module.cc:356
↓ 17 callersMethodRemove
modules/congestion_controller/bbr/packet_number_indexed_queue.h:172
↓ 17 callersMethodRemove
static
rtc_base/thread.cc:132
↓ 17 callersMethodRemoveSendStream
media/engine/webrtc_voice_engine.cc:1802
↓ 17 callersMethodReset
p2p/base/port_unittest.cc:229
↓ 17 callersFunctionRunBitExactnessTest
modules/audio_processing/gain_control_unittest.cc:63
↓ 17 callersMethodSet
rtc_base/thread_unittest.cc:320
↓ 17 callersMethodSetBitrate
api/video/video_bitrate_allocation.cc:24
↓ 17 callersMethodSetCumulativeLost
modules/rtp_rtcp/source/rtcp_packet/report_block.cc:80
↓ 17 callersMethodSetFirstPacketInSubFrame
modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h:37
↓ 17 callersMethodSetName
rtc_base/thread.cc:675
↓ 17 callersMethodSetNtp
modules/rtp_rtcp/source/rtcp_packet/rrtr.h:41
↓ 17 callersMethodSetSendParameters
media/engine/webrtc_voice_engine_unittest.cc:296
↓ 17 callersFunctionStunMethodToString
api/transport/stun.cc:1118
↓ 17 callersMethodTargetLevel
modules/audio_coding/neteq/delay_manager.cc:349
↓ 17 callersFunctionTimestampsMs
modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc:55
↓ 17 callersFunctionVerifyPacket
Checks that |packet| has the attributes given in the remaining parameters.
modules/audio_coding/neteq/red_payload_splitter_unittest.cc:146
↓ 17 callersMethodWait
video/end_to_end_tests/multi_stream_tests.cc:63
↓ 17 callersFunction_GetPath
(*path_chunks)
tools_webrtc/presubmit_checks_lib/check_orphan_headers_test.py:25
↓ 17 callersMethoddata
api/array_view.h:102
↓ 17 callersMethodextended_high_seq_num
modules/rtp_rtcp/source/rtcp_packet/report_block.h:55
↓ 17 callersMethodfirst_factory
api/test/loopback_media_transport.h:93
↓ 17 callersMethodgetRotation
Rotation of the frame in degrees.
sdk/android/api/org/webrtc/VideoFrame.java:153
↓ 17 callersMethodgetStrideU
()
sdk/android/api/org/webrtc/VideoFrame.java:89
↓ 17 callersMethodgetStrideV
()
sdk/android/api/org/webrtc/VideoFrame.java:90
↓ 17 callersMethodgetType
()
sdk/android/api/org/webrtc/VideoFrame.java:112
↓ 17 callersMethodlogD
(String string)
sdk/android/api/org/webrtc/EglRenderer.java:778
↓ 17 callersMethodrtp_transport
pc/channel.h:123
↓ 17 callersMethodset_bitrate_bps
modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h:33
↓ 17 callersMethodset_extmap_allow_mixed
Determines if it's allowed to mix one- and two-byte rtp header extensions within the same rtp stream.
pc/session_description.h:612
↓ 17 callersMethodset_num_channels
modules/audio_processing/audio_buffer.cc:218
↓ 17 callersMethodset_rtcp_mux
pc/session_description.h:110
↓ 17 callersMethodsink
media/engine/fake_webrtc_call.h:93
↓ 17 callersMethodunique_requests
Gets the number of unique NACKed RTP packets.
modules/rtp_rtcp/source/rtcp_nack_stats.h:31
↓ 16 callersMethodAddEvents
logging/rtc_event_log/rtc_event_processor.h:105
↓ 16 callersMethodAddPacket
Store packet in buffer.
media/engine/unhandled_packets_buffer.cc:26
↓ 16 callersFunctionAlignedMalloc
rtc_base/memory/aligned_malloc.h:45
↓ 16 callersMethodAllocateSid
pc/data_channel.cc:64
↓ 16 callersFunctionCanonicalFourCC
media/base/video_common.cc:42
↓ 16 callersFunctionCheckDecision
modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc:41
↓ 16 callersFunctionConfigureSvc
modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc:83
↓ 16 callersFunctionCreateConfig
modules/video_coding/codecs/test/videocodec_test_libvpx.cc:58
↓ 16 callersFunctionCreateMockWavReaderFactory
modules/audio_processing/test/conversational_speech/generator_unittest.cc:95
↓ 16 callersFunctionCreateSingleOrMultiStreamEncoder
modules/audio_coding/codecs/opus/opus_unittest.cc:39
↓ 16 callersMethodEstimate
system_wrappers/source/rtp_to_ntp_estimator.cc:182
↓ 16 callersFunctionFieldTrialsStringIsValid
system_wrappers/source/field_trial.cc:73
↓ 16 callersFunctionFindCodecById
media/base/codec.h:223
↓ 16 callersMethodGenerateFrame
modules/video_coding/test/stream_generator.cc:32
↓ 16 callersFunctionGeneratePacketMask
modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc:57
↓ 16 callersFunctionGetInitialSaturationMarginDb
modules/audio_processing/agc2/agc2_common.cc:21
↓ 16 callersMethodGetMax
Returns maximum among all seen samples, in O(1) time. This isn't affected by RemoveSample().
rtc_base/numerics/running_statistics.h:118
↓ 16 callersFunctionGetStats
pc/video_track_source.h:38
↓ 16 callersMethodHasBitrate
api/video/video_bitrate_allocation.cc:45
↓ 16 callersMethodHasOneRef
Return whether the reference count is one. If the reference count is used in the conventional way, a reference count of 1 implies that the current thr
rtc_base/ref_counter.h:60
↓ 16 callersFunctionHostToNetwork32
rtc_base/byte_order.h:170
↓ 16 callersMethodIfft
Computes the inverse Fft.
modules/audio_processing/aec3/aec3_fft.h:40
↓ 16 callersMethodInsertDtmf
pc/dtmf_sender_unittest.cc:86
↓ 16 callersMethodInsertPacket
modules/rtp_rtcp/source/rtp_sequence_number_map.cc:32
↓ 16 callersMethodIntersectWith
modules/desktop_capture/desktop_region.cc:263
↓ 16 callersFunctionNetworkToHost32
rtc_base/byte_order.h:182
↓ 16 callersMethodNextSendTime
modules/pacing/pacing_controller.cc:322
↓ 16 callersFunctionNumSamples
system_wrappers/source/metrics_default_unittest.cc:27
↓ 16 callersFunctionPacketize
modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc:88
↓ 16 callersFunctionPath
test/testsupport/file_utils_unittest.cc:35
↓ 16 callersFunctionPopulateInputFrame
Populates the frame with linearly increasing sample values for each band, with a band-specific offset, in order to allow simple bitexactness verificat
modules/audio_processing/aec3/echo_canceller3_unittest.cc:39
↓ 16 callersMethodPosition
Returns the current position in the circular buffer.
modules/audio_processing/aec3/render_buffer.h:60
↓ 16 callersMethodRTT
Get RoundTripTime.
modules/rtp_rtcp/source/rtp_rtcp_impl.cc:453
↓ 16 callersMethodRate
modules/video_coding/utility/framerate_controller.cc:79
↓ 16 callersMethodRelease
p2p/base/turn_port.cc:910
↓ 16 callersMethodRemoveContentByName
pc/session_description.cc:184
↓ 16 callersMethodReportRequest
modules/rtp_rtcp/source/rtcp_nack_stats.cc:20
↓ 16 callersMethodResample
modules/backing_track/audio_resampler.cc:51
↓ 16 callersMethodReset
modules/video_coding/jitter_buffer.cc:104
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