MCPcopy Create free account

hub / github.com/anyrtcIO-Community/anyRTC-RTMP-OpenSource / functions

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

⨍ Functions 10,022◇ Types & classes 2,276
↓ 1 callersMethodawaitNewBuffer
AnyCore/lib_rtsp/liveMedia/MPEG2TransportStreamFromESSource.cpp:115
↓ 1 callersMethodbandwidth
'Bandwidth' parameter, set in the "b=" SDP line:
AnyCore/lib_rtsp/liveMedia/include/MediaSession.hh:215
↓ 1 callersMethodbandwidth_check
AnyCore/srs_librtmp/srs_librtmp.cpp:35115
↓ 1 callersMethodbaseExtSeqNumReceived
AnyCore/lib_rtsp/liveMedia/include/RTPSource.hh:191
↓ 1 callersFunctionbitrv2conj
webrtc/common_audio/fft4g.c:799
↓ 1 callersMethodbits_per_sample
webrtc/modules/audio_device/include/audio_device_defines.h:163
↓ 1 callersMethodbool_variable
Flag variables
webrtc/base/flags.h:90
↓ 1 callersFunctionbs_init
�������ƣ� �������ܣ���ʼ���ṹ�� �� ���� �� �� ֵ���޷���ֵ,void���� ˼ ·�� �� �ϣ� */
AnyCore/plydecoder.cc:49
↓ 1 callersFunctionbs_read
AnyCore/plydecoder.cc:56
↓ 1 callersFunctionbs_read1
AnyCore/plydecoder.cc:164
↓ 1 callersMethodbuffer
AnyCore/lib_rtsp/liveMedia/MPEG2TransportStreamFromESSource.cpp:45
↓ 1 callersMethodbuildAndSendPacket
AnyCore/lib_rtsp/liveMedia/MultiFramedRTPSink.cpp:165
↓ 1 callersFunctionbuildHuffmanEncodingTable
AnyCore/lib_rtsp/liveMedia/MP3InternalsHuffman.cpp:799
↓ 1 callersMethodbytesAvailable
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:74
↓ 1 callersMethodc1_create
AnyCore/srs_librtmp/srs_librtmp.cpp:26781
↓ 1 callersMethodc2_create
AnyCore/srs_librtmp/srs_librtmp.cpp:27298
↓ 1 callersMethodcache_audio
AnyCore/srs_librtmp/srs_librtmp.cpp:18532
↓ 1 callersMethodcache_video
AnyCore/srs_librtmp/srs_librtmp.cpp:18566
↓ 1 callersMethodchangeClientInputSocket
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:1323
↓ 1 callersMethodcheck
AnyCore/srs_librtmp/srs_librtmp.cpp:13146
↓ 1 callersMethodcheckConsistentLayout
Requests new layout if necessary. Returns true if layout and surface size are consistent.
Prj-Android/app/src/main/java/org/webrtc/SurfaceViewRenderer.java:419
↓ 1 callersFunctioncheckForAuxSDPLine
AnyCore/lib_rtsp/liveMedia/MPEG4VideoFileServerMediaSubsession.cpp:58
↓ 1 callersFunctioncheckForAuxSDPLine
AnyCore/lib_rtsp/liveMedia/H265VideoFileServerMediaSubsession.cpp:56
↓ 1 callersFunctioncheckForAuxSDPLine
AnyCore/lib_rtsp/liveMedia/H264VideoFileServerMediaSubsession.cpp:56
↓ 1 callersMethodcheckForAuxSDPLine1
AnyCore/lib_rtsp/liveMedia/MPEG4VideoFileServerMediaSubsession.cpp:64
↓ 1 callersMethodcheckForAuxSDPLine1
AnyCore/lib_rtsp/liveMedia/H265VideoFileServerMediaSubsession.cpp:61
↓ 1 callersMethodcheckForAuxSDPLine1
AnyCore/lib_rtsp/liveMedia/H264VideoFileServerMediaSubsession.cpp:61
↓ 1 callersMethodcheckForXingHeader
AnyCore/lib_rtsp/liveMedia/MP3StreamState.cpp:399
↓ 1 callersMethodcheckNewSSRC
AnyCore/lib_rtsp/liveMedia/RTCP.cpp:702
↓ 1 callersMethodchunkId
AnyCore/lib_rtsp/liveMedia/AVIFileSink.cpp:107
↓ 1 callersMethodclearHandler
AnyCore/lib_rtsp/BasicUsageEnvironment/BasicTaskScheduler0.cpp:198
↓ 1 callersMethodclearPIDStatusTable
AnyCore/lib_rtsp/liveMedia/MPEG2TransportStreamFramer.cpp:87
↓ 1 callersMethodclientPortNum
AnyCore/lib_rtsp/liveMedia/include/MediaSession.hh:161
↓ 1 callersMethodcloseAllClientSessionsForServerMediaSession
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:88
↓ 1 callersMethodcloseStreamSource
AnyCore/lib_rtsp/liveMedia/OnDemandServerMediaSubsession.cpp:376
↓ 1 callersMethodcodec_info
webrtc/modules/utility/source/file_recorder_impl.cc:244
↓ 1 callersMethodcompleteNewConfig
AnyCore/lib_rtsp/liveMedia/MPEG4VideoStreamFramer.cpp:146
↓ 1 callersMethodcomplex_handshake
AnyCore/srs_librtmp/srs_librtmp.cpp:23060
↓ 1 callersMethodcomputeAbsDonFromDON
AnyCore/lib_rtsp/liveMedia/H265VideoRTPSource.cpp:149
↓ 1 callersFunctioncomputeJPEGHeaderSize
AnyCore/lib_rtsp/liveMedia/JPEGVideoRTPSource.cpp:172
↓ 1 callersFunctioncomputeSCRTimeCode
AnyCore/lib_rtsp/liveMedia/MPEG1or2FileServerDemux.cpp:255
↓ 1 callersMethodconfigStr
AnyCore/lib_rtsp/liveMedia/include/ADTSAudioFileSource.hh:35
↓ 1 callersMethodconnect
AnyCore/srs_librtmp/srs_librtmp.cpp:34906
↓ 1 callersMethodconnect_app
AnyCore/srs_librtmp/srs_librtmp.cpp:23076
↓ 1 callersMethodconnect_app2
AnyCore/srs_librtmp/srs_librtmp.cpp:23091
↓ 1 callersMethodconnectionHandler1
AnyCore/lib_rtsp/liveMedia/RTSPClient.cpp:1396
↓ 1 callersMethodcontinueAfterDESCRIBE
AnyCore/lib_rtsp/liveMedia/ProxyServerMediaSession.cpp:127
↓ 1 callersMethodcontinueAfterSETUP
AnyCore/lib_rtsp/liveMedia/ProxyServerMediaSession.cpp:293
↓ 1 callersMethodcontinueHandlingREGISTER1
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:1344
↓ 1 callersMethodcontinuePlaying
AnyCore/lib_rtsp/liveMedia/AVIFileSink.cpp:234
↓ 1 callersMethodcontinuePlaying1
AnyCore/lib_rtsp/liveMedia/BasicUDPSink.cpp:50
↓ 1 callersMethodconvert
(ByteBuffer buf, int width, int height, int stride, int textureId, float [] transformMatrix)
Prj-Android/app/src/main/java/org/webrtc/SurfaceTextureHelper.java:168
↓ 1 callersMethodconvertFramerates
(List<int[]> arrayRanges)
Prj-Android/app/src/main/java/org/webrtc/Camera1Enumerator.java:140
↓ 1 callersMethodconvertScalingTypeToVisibleFraction
Each scaling type has a one-to-one correspondence to a numeric minimum fraction of the video that must remain visible.
Prj-Android/app/src/main/java/org/webrtc/RendererCommon.java:216
↓ 1 callersMethodconvertToRTPTimestamp
AnyCore/lib_rtsp/liveMedia/RTPSink.cpp:74
↓ 1 callersMethodcopy_shared_key
AnyCore/srs_librtmp/srs_librtmp.cpp:26519
↓ 1 callersMethodcreate
(String name, CameraEventsHandler eventsHandler)
Prj-Android/app/src/main/java/org/webrtc/VideoCapturerAndroid.java:101
↓ 1 callersMethodcreateEglContext
( Context sharedContext, EGLDisplay eglDisplay, EGLConfig eglConfig)
Prj-Android/app/src/main/java/org/webrtc/EglBase10.java:284
↓ 1 callersMethodcreateEglContext
( EglBase14.Context sharedContext, EGLDisplay eglDisplay, EGLConfig eglConfig)
Prj-Android/app/src/main/java/org/webrtc/EglBase14.java:239
↓ 1 callersMethodcreateEventTrigger
AnyCore/lib_rtsp/BasicUsageEnvironment/BasicTaskScheduler0.cpp:89
↓ 1 callersFunctioncreateJPEGHeader
AnyCore/lib_rtsp/liveMedia/JPEGVideoRTPSource.cpp:180
↓ 1 callersMethodcreateNewClientConnection
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:2089
↓ 1 callersMethodcreateNewClientSession
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:2094
↓ 1 callersMethodcreateNewPresentationTimeSubsessionNormalizer
AnyCore/lib_rtsp/liveMedia/ProxyServerMediaSession.cpp:694
↓ 1 callersMethodcreatePbufferSurface
(int width, int height)
Prj-Android/app/src/main/java/org/webrtc/EglBase10.java:152
↓ 1 callersMethodcreatePbufferSurface
(int width, int height)
Prj-Android/app/src/main/java/org/webrtc/EglBase14.java:94
↓ 1 callersFunctioncreateRangeString
AnyCore/lib_rtsp/liveMedia/RTSPClient.cpp:555
↓ 1 callersFunctioncreateScaleString
AnyCore/lib_rtsp/liveMedia/RTSPClient.cpp:542
↓ 1 callersMethodcreate_payload
AnyCore/srs_librtmp/srs_librtmp.cpp:13048
↓ 1 callersMethodcreate_stream
AnyCore/srs_librtmp/srs_librtmp.cpp:23192
↓ 1 callersMethoddata
webrtc/base/array_view.h:106
↓ 1 callersMethoddata
webrtc/base/virtualsocketserver.cc:77
↓ 1 callersMethoddata
webrtc/base/trace_event.h:731
↓ 1 callersMethoddataEnd
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:72
↓ 1 callersMethoddataEnd
AnyCore/lib_rtsp/liveMedia/AVIFileSink.cpp:43
↓ 1 callersMethoddataStart
AnyCore/lib_rtsp/liveMedia/AVIFileSink.cpp:42
↓ 1 callersMethoddate
AnyCore/srs_librtmp/srs_librtmp.cpp:19368
↓ 1 callersMethoddeInitiate
AnyCore/lib_rtsp/liveMedia/MediaSession.cpp:850
↓ 1 callersMethoddeactivateStreamReplica
AnyCore/lib_rtsp/liveMedia/StreamReplicator.cpp:106
↓ 1 callersFunctiondecode
webrtc/base/stringencode.cc:92
↓ 1 callersFunctiondecodeURL
AnyCore/lib_rtsp/liveMedia/RTSPCommon.cpp:34
↓ 1 callersMethoddecrementReferenceCount
AnyCore/lib_rtsp/liveMedia/include/ServerMediaSession.hh:71
↓ 1 callersMethoddeleteEventTrigger
AnyCore/lib_rtsp/BasicUsageEnvironment/BasicTaskScheduler0.cpp:114
↓ 1 callersMethoddeliverBufferToClient
AnyCore/lib_rtsp/liveMedia/MPEG2TransportStreamFromESSource.cpp:197
↓ 1 callersMethoddeliverFrame
AnyCore/lib_rtsp/liveMedia/DeviceSource.cpp:100
↓ 1 callersMethoddeliverIncomingFrame
AnyCore/lib_rtsp/liveMedia/QCELPAudioRTPSource.cpp:396
↓ 1 callersMethoddeliverIncomingFrame
AnyCore/lib_rtsp/liveMedia/AMRAudioRTPSource.cpp:526
↓ 1 callersMethoddequeueOutputBuffer
(int dequeueTimeoutMs)
Prj-Android/app/src/main/java/org/webrtc/MediaCodecVideoDecoder.java:565
↓ 1 callersMethoddequeueTextureBuffer
(int timeoutMs)
Prj-Android/app/src/main/java/org/webrtc/MediaCodecVideoDecoder.java:531
↓ 1 callersMethodderegisterForMultiplexedRTCPPackets
AnyCore/lib_rtsp/liveMedia/include/RTPSource.hh:55
↓ 1 callersMethodderegisterRTPInterface
AnyCore/lib_rtsp/liveMedia/RTPInterface.cpp:455
↓ 1 callersMethoddetachCurrent
()
Prj-Android/app/src/main/java/org/webrtc/EglBase10.java:231
↓ 1 callersMethoddetachCurrent
()
Prj-Android/app/src/main/java/org/webrtc/EglBase14.java:174
↓ 1 callersMethoddeviceIsBlacklistedForOpenSLESUsage
()
webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java:190
↓ 1 callersMethoddeviceIsBlacklistedForOpenSLESUsage
()
Prj-Android/app/src/main/java/org/webrtc/voiceengine/WebRtcAudioUtils.java:184
↓ 1 callersMethoddisconnect
AnyCore/srs_librtmp/srs_librtmp.cpp:34912
↓ 1 callersMethoddoGetNextFrame
AnyCore/lib_rtsp/liveMedia/MP3ADUinterleaving.cpp:132
↓ 1 callersMethoddoGetNextFrame
AnyCore/lib_rtsp/liveMedia/MP3ADU.cpp:153
↓ 1 callersMethoddoReadFromFile
AnyCore/lib_rtsp/liveMedia/ByteStreamFileSource.cpp:129
↓ 1 callersMethoddoReadFromFile
AnyCore/lib_rtsp/liveMedia/WAVAudioFileSource.cpp:260
← previousnext →2,201–2,300 of 10,022, ranked by callers