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Functions10,022 in github.com/anyrtcIO-Community/anyRTC-RTMP-OpenSource

↓ 5 callersFunctionTimeSince
The number of milliseconds that have elapsed since 'earlier'.
webrtc/base/timeutils.h:86
↓ 5 callersFunctionTimeUntil
The number of milliseconds that will elapse between now and 'later'.
webrtc/base/timeutils.h:91
↓ 5 callersFunctionWebRtcSpl_DotProdIntToInt
compute two inner-products and store them to output array
webrtc/common_audio/signal_processing/resample.c:312
↓ 5 callersFunctionWebRtcSpl_DotProdIntToShort
compute two inner-products and store them to output array
webrtc/common_audio/signal_processing/resample.c:358
↓ 5 callersFunctionWebRtcSpl_UpBy2ShortToInt
interpolator input: int16_t output: int32_t (normalized, not saturated) (of length len*2) state: filter state array; length = 8
webrtc/common_audio/signal_processing/resample_by_2_internal.c:200
↓ 5 callersFunctionWriteLE32
webrtc/common_audio/wav_header.cc:116
↓ 5 callersMethodaddWord
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:1199
↓ 5 callersMethodadd_sample_unit
AnyCore/srs_librtmp/srs_librtmp.cpp:14213
↓ 5 callersMethodapply_rotation
webrtc/media/base/videocapturer.h:172
↓ 5 callersMethodawaitUninterruptibly
(final CountDownLatch latch)
Prj-Android/app/src/main/java/org/webrtc/ThreadUtils.java:106
↓ 5 callersFunctionbase64Decode
AnyCore/lib_rtsp/liveMedia/Base64.cpp:40
↓ 5 callersFunctionbyteSwap
AnyCore/lib_rtsp/liveMedia/OggFileParser.cpp:121
↓ 5 callersFunctioncalculateCRC
AnyCore/lib_rtsp/liveMedia/MPEG2TransportStreamMultiplexor.cpp:433
↓ 5 callersMethodcheckIsNotReleased
()
Prj-Android/app/src/main/java/org/webrtc/EglBase10.java:198
↓ 5 callersMethodcopy
AnyCore/srs_librtmp/srs_librtmp.cpp:13195
↓ 5 callersMethodcurPacketRTPSeqNum
Note that RTP receivers will usually not need to call either of the following two functions, because RTP sequence numbers and timestamps are usually n
AnyCore/lib_rtsp/liveMedia/include/RTPSource.hh:83
↓ 5 callersMethodduration
AnyCore/lib_rtsp/liveMedia/ServerMediaSession.cpp:169
↓ 5 callersMethodfileDuration
AnyCore/lib_rtsp/liveMedia/MatroskaFile.cpp:240
↓ 5 callersMethodfileName
AnyCore/lib_rtsp/liveMedia/include/OggFile.hh:51
↓ 5 callersMethodfirstAddress
AnyCore/lib_rtsp/groupsock/NetAddress.cpp:197
↓ 5 callersMethodfolder
webrtc/base/pathutils.cc:145
↓ 5 callersFunctionfreeifaddrs
webrtc/base/ifaddrs-android.cc:209
↓ 5 callersFunctionget2Bytes
AnyCore/lib_rtsp/liveMedia/WAVAudioFileSource.cpp:95
↓ 5 callersFunctiongetBufferSize
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:355
↓ 5 callersMethodgetCameraInfo
(int index)
Prj-Android/app/src/main/java/org/webrtc/Camera1Enumerator.java:66
↓ 5 callersMethodgetHandler
Retrieve the handler that calls onTextureFrameAvailable(). This handler is valid until dispose() is called.
Prj-Android/app/src/main/java/org/webrtc/SurfaceTextureHelper.java:401
↓ 5 callersMethodget_recv_bytes
AnyCore/srs_librtmp/srs_librtmp.cpp:21382
↓ 5 callersMethodget_send_bytes
AnyCore/srs_librtmp/srs_librtmp.cpp:21387
↓ 5 callersFunctiongroupsockPriv
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:53
↓ 5 callersMethodgs
AnyCore/lib_rtsp/liveMedia/include/RTPInterface.hh:60
↓ 5 callersMethodhandleCmd_notFound
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:615
↓ 5 callersMethodhandleRead
AnyCore/lib_rtsp/liveMedia/RTPInterface.cpp:248
↓ 5 callersMethodhaveSubframes
0 means: frames do not have subframes (the default behavior)
AnyCore/lib_rtsp/liveMedia/include/MatroskaFile.hh:135
↓ 5 callersMethodheadIndex
AnyCore/lib_rtsp/liveMedia/MP3ADU.cpp:64
↓ 5 callersMethodinsertWord
AnyCore/lib_rtsp/liveMedia/MediaSink.cpp:163
↓ 5 callersMethodipv6_flags
webrtc/base/ipaddress.h:143
↓ 5 callersMethodisWideband
AnyCore/lib_rtsp/liveMedia/AMRAudioRTPSource.cpp:40
↓ 5 callersMethodis_strict_array
AnyCore/srs_librtmp/srs_librtmp.cpp:19146
↓ 5 callersMethodlookupDemuxedTrack
AnyCore/lib_rtsp/liveMedia/OggFile.cpp:263
↓ 5 callersMethodlookupServerMediaSession
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:67
↓ 5 callersFunctionlookupXYandPutBits
AnyCore/lib_rtsp/liveMedia/MP3InternalsHuffman.cpp:828
↓ 5 callersFunctionmakect
webrtc/common_audio/fft4g.c:677
↓ 5 callersMethodmaybePostOnCameraThread
(Runnable runnable)
Prj-Android/app/src/main/java/org/webrtc/VideoCapturerAndroid.java:244
↓ 5 callersMethodmultiplyMatrices
Returns new matrix with the result of a b.
Prj-Android/app/src/main/java/org/webrtc/RendererCommon.java:158
↓ 5 callersMethodonCameraError
(String errorDescription)
Prj-Android/app/src/main/java/org/webrtc/CameraVideoCapturer.java:26
↓ 5 callersFunctionour_MD5Data
AnyCore/lib_rtsp/liveMedia/ourMD5.cpp:53
↓ 5 callersFunctionparseSPropParameterSets
AnyCore/lib_rtsp/liveMedia/H264VideoRTPSource.cpp:119
↓ 5 callersMethodrtpSink
AnyCore/lib_rtsp/liveMedia/DarwinInjector.cpp:34
↓ 5 callersMethodscale
AnyCore/lib_rtsp/liveMedia/include/RTSPClient.hh:205
↓ 5 callersMethodschema
AnyCore/srs_librtmp/srs_librtmp.cpp:27006
↓ 5 callersMethodsession
AnyCore/lib_rtsp/liveMedia/include/RTSPClient.hh:198
↓ 5 callersMethodsetParamsFromHeader
AnyCore/lib_rtsp/liveMedia/MP3Internals.cpp:156
↓ 5 callersMethodset_default_local_address_provider
webrtc/base/network.h:299
↓ 5 callersMethodset_ntp_time_ms
Set capture ntp time in milliseconds.
webrtc/video_frame.h:117
↓ 5 callersFunctionsocketJoinGroup
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:431
↓ 5 callersMethodsucceeded
webrtc/modules/audio_device/win/audio_device_core_win.h:68
↓ 5 callersMethodtellg
AnyCore/srs_librtmp/srs_librtmp.cpp:15219
↓ 5 callersFunctionupdate_offset
webrtc/system_wrappers/source/spreadsortlib/spreadsort.hpp:1003
↓ 5 callersMethodv
(String tag, String message)
Prj-Android/app/src/main/java/org/webrtc/Logging.java:136
↓ 4 callersMethodAddSample
webrtc/modules/video_coding/utility/moving_average.h:40
↓ 4 callersFunctionAdjustCurrentProcessPrivilege
webrtc/base/win32regkey.cc:1076
↓ 4 callersMethodBind
webrtc/base/asyncsocket.cc:49
↓ 4 callersMethodCheck
webrtc/base/firewallsocketserver.cc:159
↓ 4 callersMethodClose
webrtc/modules/audio_device/android/audio_manager.cc:54
↓ 4 callersMethodConsumeBits
webrtc/base/bitbuffer.cc:148
↓ 4 callersMethodCopy
webrtc/base/cryptstring.cc:29
↓ 4 callersFunctionCreateRandomId
webrtc/base/helpers.cc:283
↓ 4 callersMethodCurrentThreadIsOwner
webrtc/base/criticalsection.cc:166
↓ 4 callersMethodDecoded
Provides an alternative interface that allows the decoder to specify the decode time excluding waiting time for any previous pending frame to return.
webrtc/video_decoder.h:39
↓ 4 callersFunctionDeleteGlobalRef
webrtc/modules/utility/source/helpers_android.cc:80
↓ 4 callersMethodDeliverRecordedData
webrtc/modules/audio_device/fine_audio_buffer.cc:114
↓ 4 callersMethodDestroy
webrtc/base/nethelpers.cc:114
↓ 4 callersMethodEgl
()
Prj-Android/app/src/main/java/org/anyrtc/core/AnyRTMP.java:69
↓ 4 callersFunctionFromString
webrtc/base/stringencode.h:192
↓ 4 callersMethodGetBufferSizeInMilliseconds
webrtc/modules/audio_device/include/audio_device_defines.h:190
↓ 4 callersMethodGetBytesPerBuffer
webrtc/modules/audio_device/include/audio_device_defines.h:176
↓ 4 callersMethodGetDescriptor
webrtc/base/physicalsocketserver.cc:687
↓ 4 callersMethodGetLength
webrtc/base/cryptstring.cc:15
↓ 4 callersMethodGetLocalAddress
webrtc/base/asyncsocket.cc:41
↓ 4 callersFunctionGetObjectClass
Prj-Android/jni/jni_util/jni_helpers.cc:169
↓ 4 callersMethodGetPlayoutData
webrtc/modules/audio_device/android/audio_track_jni.cc:234
↓ 4 callersMethodGetRecordAudioParameters
webrtc/modules/audio_device/include/audio_device.h:212
↓ 4 callersMethodGetRequestedEvents
webrtc/base/physicalsocketserver.cc:728
↓ 4 callersFunctionGetTemporaryFolder
Note: this method uses the convention of <temp>/<appname> for the temporary folder. Filesystem uses <temp>/<exename>. We will be migrating exclusive
webrtc/base/pathutils.h:139
↓ 4 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
webrtc/base/refcount.h:166
↓ 4 callersFunctionIPAddressPrecedence
webrtc/base/ipaddress.cc:482
↓ 4 callersMethodInit
VideoRender/pipeline_render.cpp:364
↓ 4 callersFunctionInternalUrlEncode
webrtc/base/urlencode.cc:93
↓ 4 callersFunctionInvoke
webrtc/base/thread.h:168
↓ 4 callersMethodIsAnyIP
webrtc/base/socketaddress.cc:208
↓ 4 callersMethodIsConsistent
Pre- and postcondition of all methods.
webrtc/base/copyonwritebuffer.h:282
↓ 4 callersMethodIsLowLatencyPlayoutSupported
webrtc/modules/audio_device/android/audio_manager.cc:200
↓ 4 callersMethodIsUnresolvedIP
webrtc/base/socketaddress.cc:221
↓ 4 callersFunctionMD5Transform
The core of the MD5 algorithm, this alters an existing MD5 hash to reflect the addition of 16 longwords of new data. MD5Update blocks the data and co
webrtc/base/md5.cc:143
↓ 4 callersMethodMatch
webrtc/base/messagequeue.h:144
↓ 4 callersMethodOnMonitoringStatusChanged
webrtc/base/dbus.cc:393
↓ 4 callersMethodOnPreEvent
webrtc/base/physicalsocketserver.cc:732
↓ 4 callersMethodOpenFile
webrtc/system_wrappers/source/file_impl.cc:92
↓ 4 callersFunctionParseRbsp
webrtc/common_video/h264/h264_common.cc:61
↓ 4 callersMethodPlayoutIsInitialized
webrtc/modules/audio_device/android/audio_track_jni.h:74
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