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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 47 callersMethodSetPayloadType
modules/rtp_rtcp/source/rtp_packet.cc:140
↓ 47 callersFunctionSleepMs
system_wrappers/source/sleep.cc:24
↓ 47 callersFunctionToString
Dumps an AudioCodec in RFC 2327-ish format.
media/engine/webrtc_voice_engine.cc:88
↓ 47 callersMethodUnwrap
Unwrap the value and update the internal state.
modules/include/module_common_types_public.h:73
↓ 47 callersMethodheaders_size
modules/rtp_rtcp/source/rtp_packet.h:62
↓ 47 callersFunctionjlongFromPointer
Deprecated, use NativeToJavaPointer.
sdk/android/src/jni/jni_helpers.h:45
↓ 47 callersFunctionnetwork_thread
p2p/client/basic_port_allocator.h:120
↓ 47 callersMethodsignaling_state
pc/peer_connection.cc:2118
↓ 46 callersFunctionCreateSessionDescription
pc/jsep_session_description.cc:133
↓ 46 callersMethodGetRemoteAddress
rtc_base/async_socket.cc:45
↓ 46 callersFunctionParseFailed
The line starting at |line_start| of |message| is the failing line. The reason for the failure should be provided in the |description|. An example of
pc/webrtc_sdp.cc:382
↓ 46 callersMethodRate
rtc_base/rate_statistics.cc:90
↓ 46 callersFunctionSetFrameData
audio/utility/audio_frame_operations_unittest.cc:30
↓ 46 callersMethodbuilder
()
sdk/android/api/org/webrtc/EncodedImage.java:127
↓ 46 callersFunctionhex_encode
Convert an unsigned value from 0 to 15 to the hex character equivalent...
rtc_base/string_encode.cc:28
↓ 46 callersMethodis_open
Some helper methods to improve code readability.
media/sctp/sctp_transport.h:242
↓ 46 callersMethodnum_rendered_frames
media/base/fake_video_renderer.h:54
↓ 46 callersMethodok
logging/rtc_event_log/rtc_event_log_parser.h:269
↓ 46 callersMethodpayload_size_bytes
modules/rtp_rtcp/source/rtcp_packet/common_header.h:33
↓ 45 callersFunctionAddDataSection
pc/media_session_unittest.cc:338
↓ 45 callersMethodChannels
modules/audio_coding/neteq/audio_multi_vector.cc:186
↓ 45 callersMethodClear
static
rtc_base/thread.cc:149
↓ 45 callersMethodGetBest
modules/congestion_controller/bbr/windowed_filter.h:148
↓ 45 callersMethodInitEncode
TODO(nisse): Valid overrides commented out, because the gmock methods don't use any override declarations, and we want to avoid warnings from -Wincons
media/engine/simulcast_encoder_adapter_unittest.cc:209
↓ 45 callersMethodIsUnresolvedIP
rtc_base/socket_address.cc:216
↓ 45 callersMethodNextFrameConfig
modules/video_coding/codecs/vp8/screenshare_layers.cc:101
↓ 45 callersMethodPrepareAddress
p2p/base/port_unittest.cc:171
↓ 45 callersMethodTestMemberIsDefined
pc/rtc_stats_integrationtest.cc:221
↓ 45 callersFunctionTimeNanos
rtc_base/time_utils.cc:203
↓ 45 callersFunctionWebRtcSpl_DivW32W16
common_audio/signal_processing/division_operations.c:39
↓ 45 callersFunctionbegin
api/rtp_packet_infos.h:68
↓ 45 callersMethodreceiver
pc/rtp_transceiver.cc:249
↓ 45 callersMethodsource_ssrc
modules/rtp_rtcp/source/rtcp_packet/report_block.h:50
↓ 44 callersMethodAdd
video/stats_counter.cc:239
↓ 44 callersMethodClear
pc/data_channel.cc:123
↓ 44 callersFunctionCreateTestFixture
video/pc_full_stack_tests.cc:78
↓ 44 callersFunctionCreateTwoNetworkLinks
video/pc_full_stack_tests.cc:60
↓ 44 callersMethodInsert
rtc_base/numerics/percentile_filter.h:68
↓ 44 callersFunctionIsConsistent
Pre- and postcondition of all methods.
rtc_base/copy_on_write_buffer.h:270
↓ 44 callersFunctionIsNewerSequenceNumber
NB: Doesn't fulfill strict weak ordering requirements. Mustn't be used as std::map Compare function.
modules/include/module_common_types_public.h:88
↓ 44 callersMethodPutRtpPacket
modules/rtp_rtcp/source/rtp_packet_history.cc:121
↓ 44 callersFunctionToString
This function is declared inline because it is not clear which .cc file it should belong to. TODO(hta): Refactor. https://bugs.webrtc.org/6842 TODO(jo
modules/video_coding/codecs/h264/include/h264_globals.h:49
↓ 44 callersMethodchannel
modules/audio_processing/include/audio_frame_view.h:46
↓ 44 callersMethoddispose
Perform any final cleanup here. No more capturing will be done after this call.
sdk/android/api/org/webrtc/VideoCapturer.java:47
↓ 44 callersMethodreceiver_report
test/rtcp_packet_parser.h:93
↓ 43 callersMethodAddGroup
Group mutators.
pc/session_description.h:595
↓ 43 callersMethodConsumeBits
rtc_base/bit_buffer.cc:152
↓ 43 callersFunctionCreateRandomString
rtc_base/helpers.cc:116
↓ 43 callersFunctionGet
Returns non-owning pointer to the task queue implementation.
rtc_base/task_queue.h:90
↓ 43 callersMethodGetBestIP
Select the best IP address to use from this Network.
rtc_base/network.cc:1021
↓ 43 callersFunctionNativeToJavaString
sdk/android/native_api/jni/java_types.cc:194
↓ 43 callersMethodOnSendEncodedImage
video/send_statistics_proxy.cc:899
↓ 43 callersFunctionTo16u
Utility method for truncating sequence numbers to uint16.
modules/rtp_rtcp/source/rtp_packet_history_unittest.cc:28
↓ 43 callersMethodcalled
pc/rtc_stats_collector_unittest.cc:2889
↓ 43 callersMethodcertificate
rtc_base/rtc_certificate_generator_unittest.cc:38
↓ 43 callersMethodcheckIsOnValidThread
()
rtc_base/java/src/org/webrtc/ThreadUtils.java:28
↓ 43 callersMethodfirst_ssrc
pc/session_description.h:191
↓ 43 callersMethodlast_sent_packet
media/engine/fake_webrtc_call.h:299
↓ 42 callersFunctionInitAttrLine
Init |os| to "a=|attribute|".
pc/webrtc_sdp.cc:522
↓ 42 callersFunctionProcess
(filepath, args)
tools_webrtc/clang_tidy.py:36
↓ 42 callersMethodReleaseFrame
modules/video_coding/receiver.cc:179
↓ 42 callersMethodReset
modules/audio_coding/neteq/expand.cc:60
↓ 42 callersMethodSetLocalContent
pc/channel.cc:272
↓ 42 callersMethodSsrc
modules/rtp_rtcp/source/rtp_packet.h:59
↓ 42 callersMethodSsrc
video/end_to_end_tests/multi_stream_tests.cc:46
↓ 42 callersMethodUnwrap
rtc_base/time_utils.cc:238
↓ 42 callersMethodfir
test/rtcp_packet_parser.h:89
↓ 42 callersMethodreceiving
p2p/base/dtls_transport.cc:447
↓ 42 callersMethodstart
()
sdk/android/src/java/org/webrtc/MediaCodecWrapper.java:28
↓ 41 callersMethodAddContent
pc/session_description.cc:139
↓ 41 callersMethodFrameType
modules/video_coding/frame_buffer.cc:31
↓ 41 callersMethodOnRenderedFrame
video/video_quality_observer.cc:134
↓ 41 callersMethodOpen
modules/audio_coding/test/RTPFile.cc:134
↓ 41 callersMethodSetRemoteContent
pc/channel.cc:281
↓ 41 callersFunctionWEBRTC_SPL_MUL_16_32_RSFT16
common_audio/signal_processing/include/spl_inl_mips.h:38
↓ 41 callersMethoddata
video/frame_encode_metadata_writer.cc:34
↓ 41 callersMethodice_transport
pc/dtls_transport.cc:79
↓ 41 callersMethodsession_id
pc/peer_connection.h:255
↓ 41 callersMethodstream_ids
pc/video_rtp_receiver.cc:74
↓ 40 callersFunctionClipNameToClipPath
video/full_stack_tests.cc:78
↓ 40 callersMethodConnect
rtc_base/async_socket.cc:53
↓ 40 callersFunctionFieldTrialStructMember
rtc_base/experiments/field_trial_list.h:152
↓ 40 callersMethodParse
Picture loss indication (PLI) (RFC 4585). FCI: no feedback control information.
modules/rtp_rtcp/source/rtcp_packet/pli.cc:45
↓ 40 callersMethodProcessMessages
rtc_base/thread.cc:1007
↓ 40 callersMethodRemoveObserver
call/bitrate_allocator.cc:529
↓ 40 callersFunctionSdpSerialize
pc/webrtc_sdp.cc:857
↓ 40 callersMethodSetOptions
video/overuse_frame_detector.cc:686
↓ 40 callersMethodStart
rtc_base/thread.cc:689
↓ 40 callersMethodTriggerQualityLow
video/video_stream_encoder_unittest.cc:195
↓ 40 callersMethodUpdateMeasurements
system_wrappers/source/rtp_to_ntp_estimator.cc:122
↓ 40 callersMethodadd
api/jsep_ice_candidate.cc:46
↓ 40 callersMethodcompare
(CaptureFormat first, CaptureFormat second)
examples/androidapp/src/org/appspot/apprtc/CaptureQualityController.java:45
↓ 40 callersMethodntp_time_ms
media/base/fake_video_renderer.h:64
↓ 40 callersMethodnum_frames
modules/audio_processing/include/audio_processing.h:773
↓ 40 callersMethodpost
(String message)
examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java:164
↓ 40 callersMethodsize
rtc_base/buffer_unittest.cc:100
↓ 40 callersMethodssrc
pc/rtp_sender.h:107
↓ 40 callersMethodwidth
test/fake_texture_frame.cc:37
↓ 39 callersMethodCaptureFrame
modules/desktop_capture/window_capturer_win.cc:260
↓ 39 callersFunctionCreateConfigWithParameters
modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc:40
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