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github.com/anyrtcIO-Community/anyRTC-RTMP-OpenSource
/ functions
Functions
10,022 in github.com/anyrtcIO-Community/anyRTC-RTMP-OpenSource
⨍
Functions
10,022
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Types & classes
2,276
↓ 5 callers
Function
TimeSince
The number of milliseconds that have elapsed since 'earlier'.
webrtc/base/timeutils.h:86
↓ 5 callers
Function
TimeUntil
The number of milliseconds that will elapse between now and 'later'.
webrtc/base/timeutils.h:91
↓ 5 callers
Function
WebRtcSpl_DotProdIntToInt
compute two inner-products and store them to output array
webrtc/common_audio/signal_processing/resample.c:312
↓ 5 callers
Function
WebRtcSpl_DotProdIntToShort
compute two inner-products and store them to output array
webrtc/common_audio/signal_processing/resample.c:358
↓ 5 callers
Function
WebRtcSpl_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 callers
Function
WriteLE32
webrtc/common_audio/wav_header.cc:116
↓ 5 callers
Method
addWord
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:1199
↓ 5 callers
Method
add_sample_unit
AnyCore/srs_librtmp/srs_librtmp.cpp:14213
↓ 5 callers
Method
apply_rotation
webrtc/media/base/videocapturer.h:172
↓ 5 callers
Method
awaitUninterruptibly
(final CountDownLatch latch)
Prj-Android/app/src/main/java/org/webrtc/ThreadUtils.java:106
↓ 5 callers
Function
base64Decode
AnyCore/lib_rtsp/liveMedia/Base64.cpp:40
↓ 5 callers
Function
byteSwap
AnyCore/lib_rtsp/liveMedia/OggFileParser.cpp:121
↓ 5 callers
Function
calculateCRC
AnyCore/lib_rtsp/liveMedia/MPEG2TransportStreamMultiplexor.cpp:433
↓ 5 callers
Method
checkIsNotReleased
()
Prj-Android/app/src/main/java/org/webrtc/EglBase10.java:198
↓ 5 callers
Method
copy
AnyCore/srs_librtmp/srs_librtmp.cpp:13195
↓ 5 callers
Method
curPacketRTPSeqNum
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 callers
Method
duration
AnyCore/lib_rtsp/liveMedia/ServerMediaSession.cpp:169
↓ 5 callers
Method
fileDuration
AnyCore/lib_rtsp/liveMedia/MatroskaFile.cpp:240
↓ 5 callers
Method
fileName
AnyCore/lib_rtsp/liveMedia/include/OggFile.hh:51
↓ 5 callers
Method
firstAddress
AnyCore/lib_rtsp/groupsock/NetAddress.cpp:197
↓ 5 callers
Method
folder
webrtc/base/pathutils.cc:145
↓ 5 callers
Function
freeifaddrs
webrtc/base/ifaddrs-android.cc:209
↓ 5 callers
Function
get2Bytes
AnyCore/lib_rtsp/liveMedia/WAVAudioFileSource.cpp:95
↓ 5 callers
Function
getBufferSize
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:355
↓ 5 callers
Method
getCameraInfo
(int index)
Prj-Android/app/src/main/java/org/webrtc/Camera1Enumerator.java:66
↓ 5 callers
Method
getHandler
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 callers
Method
get_recv_bytes
AnyCore/srs_librtmp/srs_librtmp.cpp:21382
↓ 5 callers
Method
get_send_bytes
AnyCore/srs_librtmp/srs_librtmp.cpp:21387
↓ 5 callers
Function
groupsockPriv
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:53
↓ 5 callers
Method
gs
AnyCore/lib_rtsp/liveMedia/include/RTPInterface.hh:60
↓ 5 callers
Method
handleCmd_notFound
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:615
↓ 5 callers
Method
handleRead
AnyCore/lib_rtsp/liveMedia/RTPInterface.cpp:248
↓ 5 callers
Method
haveSubframes
0 means: frames do not have subframes (the default behavior)
AnyCore/lib_rtsp/liveMedia/include/MatroskaFile.hh:135
↓ 5 callers
Method
headIndex
AnyCore/lib_rtsp/liveMedia/MP3ADU.cpp:64
↓ 5 callers
Method
insertWord
AnyCore/lib_rtsp/liveMedia/MediaSink.cpp:163
↓ 5 callers
Method
ipv6_flags
webrtc/base/ipaddress.h:143
↓ 5 callers
Method
isWideband
AnyCore/lib_rtsp/liveMedia/AMRAudioRTPSource.cpp:40
↓ 5 callers
Method
is_strict_array
AnyCore/srs_librtmp/srs_librtmp.cpp:19146
↓ 5 callers
Method
lookupDemuxedTrack
AnyCore/lib_rtsp/liveMedia/OggFile.cpp:263
↓ 5 callers
Method
lookupServerMediaSession
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:67
↓ 5 callers
Function
lookupXYandPutBits
AnyCore/lib_rtsp/liveMedia/MP3InternalsHuffman.cpp:828
↓ 5 callers
Function
makect
webrtc/common_audio/fft4g.c:677
↓ 5 callers
Method
maybePostOnCameraThread
(Runnable runnable)
Prj-Android/app/src/main/java/org/webrtc/VideoCapturerAndroid.java:244
↓ 5 callers
Method
multiplyMatrices
Returns new matrix with the result of a b.
Prj-Android/app/src/main/java/org/webrtc/RendererCommon.java:158
↓ 5 callers
Method
onCameraError
(String errorDescription)
Prj-Android/app/src/main/java/org/webrtc/CameraVideoCapturer.java:26
↓ 5 callers
Function
our_MD5Data
AnyCore/lib_rtsp/liveMedia/ourMD5.cpp:53
↓ 5 callers
Function
parseSPropParameterSets
AnyCore/lib_rtsp/liveMedia/H264VideoRTPSource.cpp:119
↓ 5 callers
Method
rtpSink
AnyCore/lib_rtsp/liveMedia/DarwinInjector.cpp:34
↓ 5 callers
Method
scale
AnyCore/lib_rtsp/liveMedia/include/RTSPClient.hh:205
↓ 5 callers
Method
schema
AnyCore/srs_librtmp/srs_librtmp.cpp:27006
↓ 5 callers
Method
session
AnyCore/lib_rtsp/liveMedia/include/RTSPClient.hh:198
↓ 5 callers
Method
setParamsFromHeader
AnyCore/lib_rtsp/liveMedia/MP3Internals.cpp:156
↓ 5 callers
Method
set_default_local_address_provider
webrtc/base/network.h:299
↓ 5 callers
Method
set_ntp_time_ms
Set capture ntp time in milliseconds.
webrtc/video_frame.h:117
↓ 5 callers
Function
socketJoinGroup
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:431
↓ 5 callers
Method
succeeded
webrtc/modules/audio_device/win/audio_device_core_win.h:68
↓ 5 callers
Method
tellg
AnyCore/srs_librtmp/srs_librtmp.cpp:15219
↓ 5 callers
Function
update_offset
webrtc/system_wrappers/source/spreadsortlib/spreadsort.hpp:1003
↓ 5 callers
Method
v
(String tag, String message)
Prj-Android/app/src/main/java/org/webrtc/Logging.java:136
↓ 4 callers
Method
AddSample
webrtc/modules/video_coding/utility/moving_average.h:40
↓ 4 callers
Function
AdjustCurrentProcessPrivilege
webrtc/base/win32regkey.cc:1076
↓ 4 callers
Method
Bind
webrtc/base/asyncsocket.cc:49
↓ 4 callers
Method
Check
webrtc/base/firewallsocketserver.cc:159
↓ 4 callers
Method
Close
webrtc/modules/audio_device/android/audio_manager.cc:54
↓ 4 callers
Method
ConsumeBits
webrtc/base/bitbuffer.cc:148
↓ 4 callers
Method
Copy
webrtc/base/cryptstring.cc:29
↓ 4 callers
Function
CreateRandomId
webrtc/base/helpers.cc:283
↓ 4 callers
Method
CurrentThreadIsOwner
webrtc/base/criticalsection.cc:166
↓ 4 callers
Method
Decoded
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 callers
Function
DeleteGlobalRef
webrtc/modules/utility/source/helpers_android.cc:80
↓ 4 callers
Method
DeliverRecordedData
webrtc/modules/audio_device/fine_audio_buffer.cc:114
↓ 4 callers
Method
Destroy
webrtc/base/nethelpers.cc:114
↓ 4 callers
Method
Egl
()
Prj-Android/app/src/main/java/org/anyrtc/core/AnyRTMP.java:69
↓ 4 callers
Function
FromString
webrtc/base/stringencode.h:192
↓ 4 callers
Method
GetBufferSizeInMilliseconds
webrtc/modules/audio_device/include/audio_device_defines.h:190
↓ 4 callers
Method
GetBytesPerBuffer
webrtc/modules/audio_device/include/audio_device_defines.h:176
↓ 4 callers
Method
GetDescriptor
webrtc/base/physicalsocketserver.cc:687
↓ 4 callers
Method
GetLength
webrtc/base/cryptstring.cc:15
↓ 4 callers
Method
GetLocalAddress
webrtc/base/asyncsocket.cc:41
↓ 4 callers
Function
GetObjectClass
Prj-Android/jni/jni_util/jni_helpers.cc:169
↓ 4 callers
Method
GetPlayoutData
webrtc/modules/audio_device/android/audio_track_jni.cc:234
↓ 4 callers
Method
GetRecordAudioParameters
webrtc/modules/audio_device/include/audio_device.h:212
↓ 4 callers
Method
GetRequestedEvents
webrtc/base/physicalsocketserver.cc:728
↓ 4 callers
Function
GetTemporaryFolder
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 callers
Method
HasOneRef
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 callers
Function
IPAddressPrecedence
webrtc/base/ipaddress.cc:482
↓ 4 callers
Method
Init
VideoRender/pipeline_render.cpp:364
↓ 4 callers
Function
InternalUrlEncode
webrtc/base/urlencode.cc:93
↓ 4 callers
Function
Invoke
webrtc/base/thread.h:168
↓ 4 callers
Method
IsAnyIP
webrtc/base/socketaddress.cc:208
↓ 4 callers
Method
IsConsistent
Pre- and postcondition of all methods.
webrtc/base/copyonwritebuffer.h:282
↓ 4 callers
Method
IsLowLatencyPlayoutSupported
webrtc/modules/audio_device/android/audio_manager.cc:200
↓ 4 callers
Method
IsUnresolvedIP
webrtc/base/socketaddress.cc:221
↓ 4 callers
Function
MD5Transform
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 callers
Method
Match
webrtc/base/messagequeue.h:144
↓ 4 callers
Method
OnMonitoringStatusChanged
webrtc/base/dbus.cc:393
↓ 4 callers
Method
OnPreEvent
webrtc/base/physicalsocketserver.cc:732
↓ 4 callers
Method
OpenFile
webrtc/system_wrappers/source/file_impl.cc:92
↓ 4 callers
Function
ParseRbsp
webrtc/common_video/h264/h264_common.cc:61
↓ 4 callers
Method
PlayoutIsInitialized
webrtc/modules/audio_device/android/audio_track_jni.h:74
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