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

⨍ Functions 10,022◇ Types & classes 2,276
↓ 9 callersMethodtotal_size
AnyCore/srs_librtmp/srs_librtmp.cpp:19593
↓ 9 callersMethodttl
AnyCore/lib_rtsp/groupsock/include/GroupEId.hh:41
↓ 8 callersMethodAppendFolder
webrtc/base/pathutils.cc:181
↓ 8 callersFunctionCloseInputFile
AnyCore/lib_rtsp/liveMedia/InputFile.cpp:43
↓ 8 callersMethodHandleCallbacks
Set default actions of the mock object. We are delegating to fake implementations (of AudioStreamInterface) here.
webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc:395
↓ 8 callersFunctionIsCurrent
webrtc/base/thread.h:126
↓ 8 callersFunctionIsMulticastAddress
AnyCore/lib_rtsp/groupsock/NetAddress.cpp:280
↓ 8 callersFunctionIsThreadRefEqual
webrtc/base/platform_thread.cc:50
↓ 8 callersMethodMIMEtype
AnyCore/lib_rtsp/liveMedia/MP3ADU.cpp:127
↓ 8 callersFunctionOpenOutputFile
AnyCore/lib_rtsp/liveMedia/OutputFile.cpp:32
↓ 8 callersMethodReleaseLockExclusive
webrtc/system_wrappers/source/rw_lock_win.cc:50
↓ 8 callersMethodSSRC
AnyCore/lib_rtsp/liveMedia/include/RTPSink.hh:102
↓ 8 callersFunctionSTLIsSorted
webrtc/system_wrappers/include/stl_util.h:208
↓ 8 callersMethodStop
webrtc/modules/utility/source/process_thread_impl.cc:84
↓ 8 callersMethodToString
webrtc/base/network.cc:978
↓ 8 callersFunctionWebRtcSpl_DownsampleBy2
webrtc/common_audio/signal_processing/resample_by_2.c:70
↓ 8 callersFunctionWebRtcSpl_NormW32
Return the number of steps a can be left-shifted without overflow, or 0 if a == 0.
webrtc/common_audio/signal_processing/include/spl_inl.h:130
↓ 8 callersFunction_stricmp
webrtc/base/stringutils.h:112
↓ 8 callersFunctionbase64Encode
AnyCore/lib_rtsp/liveMedia/Base64.cpp:88
↓ 8 callersMethodensure_property_string
AnyCore/srs_librtmp/srs_librtmp.cpp:19525
↓ 8 callersMethodframes_per_10ms_buffer
webrtc/modules/audio_device/include/audio_device_defines.h:174
↓ 8 callersFunctiongetNextSample
AnyCore/lib_rtsp/liveMedia/MP3InternalsHuffman.cpp:695
↓ 8 callersMethodgetThreadInfo
()
webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java:178
↓ 8 callersMethodgetThreadInfo
()
Prj-Android/app/src/main/java/org/webrtc/voiceengine/WebRtcAudioUtils.java:172
↓ 8 callersMethodhandleClosure
AnyCore/lib_rtsp/liveMedia/FramedSource.cpp:95
↓ 8 callersMethodheight
webrtc/common_video/video_frame_buffer.cc:97
↓ 8 callersFunctionhex_val
AnyCore/srs_librtmp/srs_librtmp.cpp:31692
↓ 8 callersMethodisEmpty
AnyCore/lib_rtsp/liveMedia/MP3ADU.cpp:69
↓ 8 callersMethodmaxSize
AnyCore/lib_rtsp/liveMedia/MatroskaDemuxedTrack.hh:47
↓ 8 callersMethodnumAddresses
AnyCore/lib_rtsp/groupsock/include/NetAddress.hh:69
↓ 8 callersMethodpassword
AnyCore/lib_rtsp/liveMedia/include/DigestAuthentication.hh:54
↓ 8 callersMethodreset
AnyCore/anyrtmpull.h:39
↓ 8 callersMethodrun
()
Prj-Android/app/src/main/java/org/webrtc/ThreadUtils.java:46
↓ 8 callersMethodseconds
AnyCore/lib_rtsp/BasicUsageEnvironment/include/DelayQueue.hh:37
↓ 8 callersFunctionsrs_update_system_time_ms
AnyCore/srs_librtmp/srs_librtmp.cpp:12080
↓ 8 callersMethodto_object
AnyCore/srs_librtmp/srs_librtmp.cpp:19203
↓ 8 callersMethodwidth
webrtc/common_video/video_frame_buffer.cc:93
↓ 7 callersMethodAddIP
Adds an active IP address to this network. Does not check for duplicates.
webrtc/base/network.h:345
↓ 7 callersMethodCreateAsyncSocket
webrtc/base/macsocketserver.cc:40
↓ 7 callersMethodGetDelayEstimateInMilliseconds
webrtc/modules/audio_device/android/audio_manager.cc:218
↓ 7 callersFunctionGetMap
Gets the map (or nullptr).
webrtc/system_wrappers/source/metrics_default.cc:196
↓ 7 callersMethodGetValue
webrtc/base/win32regkey.cc:122
↓ 7 callersMethodHoster
Prj-Android/jni/jRTMPHosterImpl.h:31
↓ 7 callersFunctionI420PSNR
Compute PSNR for an I420 frame (all planes)
webrtc/common_video/libyuv/webrtc_libyuv.cc:293
↓ 7 callersMethodNewObject
webrtc/modules/utility/source/jvm_android.cc:144
↓ 7 callersMethodNext
Advances to the next file returns true if there were more files in the directory.
webrtc/base/fileutils.cc:89
↓ 7 callersFunctionPrintError
webrtc/base/checks.cc:52
↓ 7 callersFunctionPrintVideoFrame
TODO(nisse): Belongs with the test code?
webrtc/common_video/libyuv/webrtc_libyuv.cc:106
↓ 7 callersMethodRead
Read samples from file stored in memory (at construction) and copy |num_frames| (<=> 10ms) to the |destination| byte buffer.
webrtc/modules/audio_device/android/audio_device_unittest.cc:129
↓ 7 callersMethodReset
webrtc/modules/video_coding/utility/moving_average.h:61
↓ 7 callersMethodRun
webrtc/base/thread.cc:332
↓ 7 callersFunctionSetVTSessionProperty
Convenience function for setting a VT property.
webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc:59
↓ 7 callersMethodSetValue
webrtc/base/win32regkey.cc:58
↓ 7 callersMethodSetVideoCapturer
(final long renderPointer, final boolean front)
Prj-Android/app/src/main/java/org/anyrtc/core/RTMPHosterKit.java:110
↓ 7 callersFunctionTimeDiff
webrtc/base/timeutils.cc:118
↓ 7 callersFunctionTypeOfEvent
AnyCore/lib_rtsp/liveMedia/RTCP.cpp:1012
↓ 7 callersFunctionVectorToString
webrtc/media/base/mediachannel.h:78
↓ 7 callersMethodWriteAll
webrtc/base/stream.cc:44
↓ 7 callersMethodWriteNalu
webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc:355
↓ 7 callersMethodaddByte
AnyCore/lib_rtsp/liveMedia/include/AVIFileSink.hh:85
↓ 7 callersMethodaddHalfWord
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:1206
↓ 7 callersMethodauth
AnyCore/lib_rtsp/liveMedia/include/ProxyServerMediaSession.hh:53
↓ 7 callersMethodchangeDestinationParameters
AnyCore/lib_rtsp/groupsock/Groupsock.cpp:169
↓ 7 callersMethodcheckIsOnCameraThread
()
Prj-Android/app/src/main/java/org/webrtc/VideoCapturerAndroid.java:234
↓ 7 callersMethodcheckOnMediaCodecThread
()
Prj-Android/app/src/main/java/org/webrtc/MediaCodecVideoEncoder.java:305
↓ 7 callersMethodcodecName
AnyCore/lib_rtsp/liveMedia/ProxyServerMediaSession.cpp:37
↓ 7 callersFunctioncreate_json
AnyCore/srs_librtmp/srs_librtmp.cpp:31650
↓ 7 callersMethoddump
AnyCore/srs_librtmp/srs_librtmp.cpp:15861
↓ 7 callersMethodfmtp_config
AnyCore/lib_rtsp/liveMedia/MediaSession.cpp:892
↓ 7 callersMethodis_amf3_command
AnyCore/srs_librtmp/srs_librtmp.cpp:12958
↓ 7 callersFunctionmakeSocketNonBlocking
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:171
↓ 7 callersMethodnoMoreBits
AnyCore/lib_rtsp/liveMedia/OggFileParser.cpp:283
↓ 7 callersMethodntp_time_ms
Get capture ntp time in milliseconds.
webrtc/video_frame.h:122
↓ 7 callersFunctionourIPAddress
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:580
↓ 7 callersFunctionour_random32
AnyCore/lib_rtsp/groupsock/inet.c:423
↓ 7 callersMethodport
AnyCore/lib_rtsp/groupsock/include/NetInterface.hh:92
↓ 7 callersFunctionreadSocket
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:279
↓ 7 callersFunctionrftfsub
webrtc/common_audio/fft4g.c:1240
↓ 7 callersMethodrotation
Naming convention for Coordination of Video Orientation. Please see http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120
webrtc/video_frame.h:134
↓ 7 callersMethodrtpSeqNo
AnyCore/lib_rtsp/liveMedia/include/MultiFramedRTPSource.hh:110
↓ 7 callersFunctionseqNumLT
AnyCore/lib_rtsp/liveMedia/RTPSource.cpp:399
↓ 7 callersMethodset_timestamp
Set frame timestamp (90kHz).
webrtc/video_frame.h:111
↓ 7 callersFunctionshiftBits
AnyCore/lib_rtsp/liveMedia/BitVector.cpp:143
↓ 7 callersFunctionsrs_avc_startswith_annexb
AnyCore/srs_librtmp/srs_librtmp.cpp:12369
↓ 7 callersMethodsubsession
AnyCore/lib_rtsp/liveMedia/include/RTSPClient.hh:199
↓ 7 callersMethodtimeout_time
webrtc/base/task.h:112
↓ 7 callersMethodturnOnBackgroundReadHandling
The following two functions are deprecated, and are provided for backwards-compatibility only:
AnyCore/lib_rtsp/UsageEnvironment/include/UsageEnvironment.hh:158
↓ 7 callersMethoduseconds
AnyCore/lib_rtsp/BasicUsageEnvironment/include/DelayQueue.hh:43
↓ 6 callersMethodAcquireLockExclusive
webrtc/system_wrappers/source/rw_lock_win.cc:46
↓ 6 callersMethodAddRef
webrtc/base/refcount.h:150
↓ 6 callersFunctionCalcBufferSize
webrtc/common_video/libyuv/webrtc_libyuv.cc:57
↓ 6 callersFunctionCheckCrop
The offsets and sizes describe the rectangle extracted from the original (gradient) frame, in relative coordinates where the original frame correspond
webrtc/common_video/i420_video_frame_unittest.cc:56
↓ 6 callersFunctionCreateGradient
webrtc/common_video/i420_video_frame_unittest.cc:30
↓ 6 callersMethodCropAndScaleFrom
webrtc/common_video/video_frame_buffer.cc:164
↓ 6 callersMethodFlush
webrtc/modules/audio_device/android/audio_device_unittest.cc:231
↓ 6 callersMethodGetCurrentOffset
webrtc/base/bitbuffer.cc:203
↓ 6 callersFunctionGetEnv
Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
Prj-Android/jni/jni_util/jni_helpers.cc:36
↓ 6 callersMethodGetLocalAddress
webrtc/base/virtualsocketserver.cc:126
↓ 6 callersMethodGetThreadRef
webrtc/base/platform_thread.cc:162
↓ 6 callersMethodHandleCallbacks
Set default actions of the mock object. We are delegating to fake implementations (of AudioStreamInterface) here.
webrtc/modules/audio_device/android/audio_device_unittest.cc:405
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