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

⨍ Functions 10,022◇ Types & classes 2,276
↓ 23 callersFunctionstrDupSize
AnyCore/lib_rtsp/UsageEnvironment/strDup.cpp:34
↓ 22 callersMethodRemove
AnyCore/lib_rtsp/groupsock/Groupsock.cpp:595
↓ 22 callersMethodenv
webrtc/modules/utility/source/helpers_android.cc:121
↓ 22 callersMethodreadSource
AnyCore/lib_rtsp/liveMedia/include/MediaSession.hh:180
↓ 22 callersFunctionsrs_amf0_write_string
AnyCore/srs_librtmp/srs_librtmp.cpp:20603
↓ 21 callersMethodSet
webrtc/base/event.cc:41
↓ 21 callersFunctiondateHeader
AnyCore/lib_rtsp/liveMedia/RTSPCommon.cpp:343
↓ 21 callersMethodenqueueWord
AnyCore/lib_rtsp/liveMedia/MediaSink.cpp:144
↓ 21 callersFunctionsrs_amf0_read_string
AnyCore/srs_librtmp/srs_librtmp.cpp:20580
↓ 20 callersFunctionBind
webrtc/base/bind.h:161
↓ 20 callersFunctionSeekFile64
AnyCore/lib_rtsp/liveMedia/InputFile.cpp:73
↓ 20 callersMethodchannels
Returns a pointer array to the full-band channels (or lower band channels). Usage: channels()[channel][sample]. Where: 0 <= channel < |num_channels_|
webrtc/common_audio/channel_buffer.h:68
↓ 20 callersMethoddecode
AnyCore/srs_librtmp/srs_librtmp.cpp:16015
↓ 20 callersMethodequals
(Object other)
Prj-Android/app/src/main/java/org/webrtc/Size.java:31
↓ 20 callersMethodport
webrtc/base/socketaddress.cc:135
↓ 20 callersFunctionsocketErr
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:38
↓ 20 callersMethodwrite_4bytes
AnyCore/srs_librtmp/srs_librtmp.cpp:11877
↓ 19 callersMethodBytesRemaining
webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc:320
↓ 19 callersFunctionGetSymbols
webrtc/base/dbus.cc:51
↓ 19 callersMethodSetPathname
webrtc/base/pathutils.cc:122
↓ 19 callersFunctionTime
Deprecated. Do not use this in any new code.
webrtc/base/timeutils.h:67
↓ 19 callersFunctionWebRtcSpl_SatW32ToW16
webrtc/common_audio/signal_processing/include/spl_inl.h:75
↓ 19 callersFunctioncftfsub
webrtc/common_audio/fft4g.c:908
↓ 19 callersMethodheight
webrtc/media/engine/webrtcvideoframe.cc:72
↓ 19 callersMethodnumChannels
AnyCore/lib_rtsp/liveMedia/include/RTPSink.hh:49
↓ 19 callersMethodread
AnyCore/srs_librtmp/srs_librtmp.cpp:15341
↓ 19 callersMethodwidth
webrtc/media/engine/webrtcvideoframe.cc:68
↓ 18 callersMethodCallBooleanMethod
webrtc/modules/utility/source/jvm_android.cc:106
↓ 18 callersMethodCallVoidMethod
webrtc/modules/utility/source/jvm_android.cc:124
↓ 18 callersMethodGet
webrtc/modules/audio_device/android/opensles_common.h:41
↓ 18 callersFunctionMUL_ACCUM_1
webrtc/common_audio/signal_processing/resample_by_2.c:30
↓ 18 callersFunctionMUL_ACCUM_2
webrtc/common_audio/signal_processing/resample_by_2.c:46
↓ 18 callersMethodcompare
(T t1, T t2)
Prj-Android/app/src/main/java/org/webrtc/CameraEnumerationAndroid.java:170
↓ 18 callersMethoddisconnect
webrtc/base/sigslot.h:553
↓ 18 callersMethodinputSource
AnyCore/lib_rtsp/liveMedia/include/FramedFilter.hh:30
↓ 18 callersMethodrender_time_ms
Get render time in milliseconds.
webrtc/video_frame.h:145
↓ 18 callersMethodrtpSource
AnyCore/lib_rtsp/liveMedia/include/MediaSession.hh:176
↓ 18 callersMethodsignal_connect
webrtc/base/sigslot.h:442
↓ 18 callersMethodsignal_disconnect
webrtc/base/sigslot.h:448
↓ 18 callersMethodsize
webrtc/base/virtualsocketserver.cc:78
↓ 18 callersFunctionsrs_amf0_read_number
AnyCore/srs_librtmp/srs_librtmp.cpp:20684
↓ 18 callersFunctionsrs_amf0_write_number
AnyCore/srs_librtmp/srs_librtmp.cpp:20718
↓ 18 callersFunctionswap
webrtc/base/scoped_ptr.h:592
↓ 17 callersMethodemit
webrtc/base/sigslot.h:2208
↓ 17 callersMethodget1Bit
AnyCore/lib_rtsp/liveMedia/BitVector.cpp:109
↓ 17 callersMethodname
webrtc/base/flags.h:83
↓ 17 callersMethodsample_rate
webrtc/modules/audio_device/include/audio_device_defines.h:171
↓ 16 callersFunctionAttachCurrentThreadIfNeeded
Return a |JNIEnv*| usable on this thread. Attaches to |g_jvm| if necessary.
Prj-Android/jni/jni_util/jni_helpers.cc:101
↓ 16 callersMethodGetAdjustedBitrateBps
webrtc/common_video/bitrate_adjuster.cc:68
↓ 16 callersMethodGetMethodId
JavaClass implementation.
webrtc/modules/utility/source/jvm_android.cc:158
↓ 16 callersMethodadd
AnyCore/srs_librtmp/srs_librtmp.cpp:31590
↓ 16 callersMethodchannels
webrtc/modules/audio_device/include/audio_device_defines.h:172
↓ 16 callersMethodclear
AnyCore/srs_librtmp/srs_librtmp.cpp:14195
↓ 16 callersMethodgetResourceName
()
Prj-Android/app/src/main/java/org/webrtc/SurfaceViewRenderer.java:397
↓ 16 callersMethodisCurrentlyAwaitingData
AnyCore/lib_rtsp/liveMedia/include/FramedSource.hh:61
↓ 16 callersMethodread_4bytes
AnyCore/srs_librtmp/srs_librtmp.cpp:11808
↓ 16 callersMethodtype
Flag type
webrtc/base/flags.h:87
↓ 16 callersMethodwrite_2bytes
AnyCore/srs_librtmp/srs_librtmp.cpp:11868
↓ 15 callersMethodDetachFromThread
webrtc/base/thread_checker_impl.cc:33
↓ 15 callersMethodRelease
webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc:268
↓ 15 callersFunctionTimeNanos
webrtc/base/timeutils.cc:90
↓ 15 callersFunctionToUtf8
webrtc/base/win32.h:61
↓ 15 callersMethodcount
AnyCore/srs_librtmp/srs_librtmp.cpp:13140
↓ 15 callersMethoddataHere
AnyCore/lib_rtsp/liveMedia/MP3ADU.cpp:500
↓ 15 callersMethodget
webrtc/base/scoped_ptr.h:391
↓ 15 callersMethodread_bytes
AnyCore/srs_librtmp/srs_librtmp.cpp:11852
↓ 15 callersMethodsize
webrtc/base/testutils.h:398
↓ 15 callersFunctionsize_bins
webrtc/system_wrappers/source/spreadsortlib/spreadsort.hpp:107
↓ 15 callersMethodskipBits
AnyCore/lib_rtsp/liveMedia/OggFileParser.cpp:274
↓ 15 callersMethodtimestamp
Get frame timestamp (90kHz).
webrtc/video_frame.h:114
↓ 14 callersMethodClear
webrtc/base/thread.cc:458
↓ 14 callersFunctionGetJVM
Prj-Android/jni/jni_util/jni_helpers.cc:30
↓ 14 callersMethodMutableDataY
webrtc/common_video/video_frame_buffer.cc:33
↓ 14 callersMethodRegisterAudioCallback
webrtc/modules/audio_device/include/fake_audio_device.h:24
↓ 14 callersMethodRelease
webrtc/base/signalthread.cc:77
↓ 14 callersMethodRemoveNext
AnyCore/lib_rtsp/UsageEnvironment/HashTable.cpp:33
↓ 14 callersMethodSetFolder
webrtc/base/pathutils.cc:173
↓ 14 callersMethodallocated_size
TODO(nisse): Delete. Besides test code, only one use, in webrtcvideoengine2.cc:CreateBlackFrame.
webrtc/common_video/video_frame.cc:121
↓ 14 callersMethodlog
(Severity severity, String tag, String message)
webrtc/base/java/src/org/webrtc/Logging.java:103
↓ 14 callersFunctionour_random
AnyCore/lib_rtsp/groupsock/inet.c:61
↓ 14 callersMethodrtpTimestampFrequency
AnyCore/lib_rtsp/liveMedia/include/RTPSink.hh:43
↓ 14 callersFunctionsrs_avc_nalu_read_uev
AnyCore/srs_librtmp/srs_librtmp.cpp:12014
↓ 14 callersMethodto_number
AnyCore/srs_librtmp/srs_librtmp.cpp:19182
↓ 13 callersMethodCreateEmptyFrame
webrtc/common_video/video_frame.cc:45
↓ 13 callersFunctionFindClass
Returns a global reference guaranteed to be valid for the lifetime of the process.
Prj-Android/jni/jni_util/classreferenceholder.cc:100
↓ 13 callersMethodMutableDataU
webrtc/common_video/video_frame_buffer.cc:37
↓ 13 callersMethodMutableDataV
webrtc/common_video/video_frame_buffer.cc:41
↓ 13 callersMethodassign
AnyCore/lib_rtsp/groupsock/GroupEId.cpp:26
↓ 13 callersMethodcheckNoGLES2Error
(String msg)
Prj-Android/app/src/main/java/org/webrtc/GlUtil.java:26
↓ 13 callersMethodexecute
(final Runnable runnable)
Prj-Android/app/src/main/java/org/anyrtc/core/LooperExecutor.java:101
↓ 13 callersMethodgetResultMsg
AnyCore/lib_rtsp/BasicUsageEnvironment/BasicUsageEnvironment0.cpp:42
↓ 13 callersMethodlog
(Severity severity, String tag, String message)
Prj-Android/app/src/main/java/org/webrtc/Logging.java:88
↓ 13 callersMethodrealm
AnyCore/lib_rtsp/liveMedia/include/DigestAuthentication.hh:51
↓ 13 callersMethodstopGettingFrames
AnyCore/lib_rtsp/liveMedia/FramedSource.cpp:107
↓ 13 callersFunctionstrchr
webrtc/base/stringutils.h:90
↓ 13 callersMethodturnOffBackgroundReadHandling
AnyCore/lib_rtsp/UsageEnvironment/include/UsageEnvironment.hh:161
↓ 12 callersFunctionAlignedMalloc
webrtc/system_wrappers/source/aligned_malloc.cc:53
↓ 12 callersFunctionConvertToI420
TODO(nisse): Delete this wrapper, let callers use libyuv directly.
webrtc/common_video/libyuv/webrtc_libyuv.cc:244
↓ 12 callersFunctionGetFileSize
AnyCore/lib_rtsp/liveMedia/InputFile.cpp:48
↓ 12 callersMethodIsCurrent
static
webrtc/base/task_queue_win.cc:77
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