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

↓ 3 callersMethodParseBitstream
webrtc/modules/video_coding/utility/h264_bitstream_parser.cc:277
↓ 3 callersMethodPlaying
webrtc/modules/audio_device/audio_device_impl.cc:1495
↓ 3 callersMethodPlaying
webrtc/modules/audio_device/android/audio_track_jni.h:78
↓ 3 callersFunctionPointerTojlong
Return a |jlong| that will correctly convert back to |ptr|. This is needed because the alternative (of silently passing a 32-bit pointer to a vararg
webrtc/modules/utility/source/helpers_android.cc:37
↓ 3 callersFunctionPostTaskAndReply
webrtc/base/task_queue.h:195
↓ 3 callersFunctionPrintPlane
webrtc/common_video/libyuv/webrtc_libyuv.cc:93
↓ 3 callersMethodReceive
webrtc/modules/audio_device/android/opensles_common.h:34
↓ 3 callersMethodRecording
webrtc/modules/audio_device/audio_device_impl.cc:1524
↓ 3 callersMethodRecording
webrtc/modules/audio_device/android/audio_record_jni.h:82
↓ 3 callersMethodRecording
webrtc/modules/audio_device/ios/audio_device_ios.h:68
↓ 3 callersMethodRecv
webrtc/base/asyncsocket.cc:67
↓ 3 callersMethodRegisterNatives
webrtc/modules/utility/source/jvm_android.cc:186
↓ 3 callersMethodRemoveConnection
webrtc/base/virtualsocketserver.cc:737
↓ 3 callersMethodRemoveSink
webrtc/media/base/videocapturer.cc:189
↓ 3 callersMethodReportFramerate
Report framerate(fps) to estimate # of samples.
webrtc/modules/video_coding/utility/quality_scaler.cc:83
↓ 3 callersMethodReportQP
webrtc/modules/video_coding/utility/quality_scaler.cc:88
↓ 3 callersMethodResample10Msec
webrtc/modules/audio_coding/acm2/acm_resampler.cc:28
↓ 3 callersFunctionSHA1Transform
Hash a single 512-bit block. This is the core of the algorithm.
webrtc/base/sha1.cc:164
↓ 3 callersFunctionSHA1Update
Run your data through this.
webrtc/base/sha1.cc:233
↓ 3 callersMethodScaleFrom
webrtc/common_video/video_frame_buffer.cc:214
↓ 3 callersFunctionSchedule
AnyCore/lib_rtsp/liveMedia/RTCP.cpp:984
↓ 3 callersMethodSetAllowBlockingCalls
webrtc/base/thread.cc:296
↓ 3 callersMethodSetApplyRotation
webrtc/media/base/videoframefactory.h:50
↓ 3 callersMethodSetFilename
webrtc/base/pathutils.cc:224
↓ 3 callersMethodSetMessageQueue
webrtc/base/win32socketserver.cc:764
↓ 3 callersMethodSetPlayoutSampleRate
webrtc/modules/audio_device/include/fake_audio_device.h:140
↓ 3 callersMethodSetSpeakerVolume
webrtc/modules/audio_device/android/audio_track_jni.cc:173
↓ 3 callersFunctionSetTraceValue
std::string version of SetTraceValue so that trace arguments can be strings.
webrtc/base/trace_event.h:803
↓ 3 callersMethodSetVideoRender
AnyCore/anyrtmplayer.cc:72
↓ 3 callersMethodStart
webrtc/modules/utility/source/process_thread_impl.cc:61
↓ 3 callersMethodStartRtmpPlay
(final String strUrl, final long renderPointer)
Prj-Android/app/src/main/java/org/anyrtc/core/RTMPGuestKit.java:68
↓ 3 callersMethodStartRtmpStream
(final String strUrl)
Prj-Android/app/src/main/java/org/anyrtc/core/RTMPHosterKit.java:149
↓ 3 callersMethodSubtract
webrtc/common_types.h:824
↓ 3 callersMethodTerminate
webrtc/modules/audio_device/include/fake_audio_device.h:41
↓ 3 callersMethodToJava
Prj-Android/jni/jni_util/native_handle_impl.cc:35
↓ 3 callersMethodToString
webrtc/media/base/codec.cc:199
↓ 3 callersFunctionTruncateIP
webrtc/base/ipaddress.cc:337
↓ 3 callersFunctionType2String
webrtc/base/flags.cc:63
↓ 3 callersFunctionUNLOCK_FUNCTION
webrtc/base/thread_checker.h:98
↓ 3 callersFunctionUtf8ToWindowsFilename
webrtc/base/win32.cc:369
↓ 3 callersMethodVerifyAndAllocateFragmentationHeader
webrtc/modules/include/module_common_types.h:387
↓ 3 callersFunctionWebRtcSpl_AddSatW16
webrtc/common_audio/signal_processing/include/spl_inl.h:114
↓ 3 callersFunctionWebRtcSpl_CreateRealFFT
webrtc/common_audio/signal_processing/real_fft.c:21
↓ 3 callersFunctionWebRtcSpl_DivW32W16
webrtc/common_audio/signal_processing/division_operations.c:38
↓ 3 callersFunctionWebRtcSpl_GetSizeInBits
webrtc/common_audio/signal_processing/include/spl_inl.h:124
↓ 3 callersFunctionWebRtcSpl_ResampDotProduct
compute two inner-products and store them to output array
webrtc/common_audio/signal_processing/resample_fractional.c:147
↓ 3 callersFunctionWebRtcSpl_Resample22khzTo16khz
22 -> 16 resampler
webrtc/common_audio/signal_processing/resample.c:45
↓ 3 callersFunctionWebRtcSpl_ResetResample22khzTo16khz
initialize state of 22 -> 16 resampler
webrtc/common_audio/signal_processing/resample.c:96
↓ 3 callersFunctionWebRtcSpl_UpBy2IntToShort
interpolator input: int32_t (shifted 15 positions to the left, + offset 16384) output: int16_t (saturated) (of length len*2) state: filter state arr
webrtc/common_audio/signal_processing/resample_by_2_internal.c:340
↓ 3 callersFunctionWebRtc_MoveReadPtr
webrtc/common_audio/ring_buffer.c:178
↓ 3 callersFunctionWebRtc_available_write
webrtc/common_audio/ring_buffer.c:227
↓ 3 callersFunction_srs_expect_bandwidth_packet
AnyCore/srs_librtmp/srs_librtmp.cpp:35054
↓ 3 callersMethodactive
When we enumerate networks and find a previously-seen network is missing, we do not remove it (because it may be used elsewhere). Instead, we mark it
webrtc/base/network.h:399
↓ 3 callersMethodaddress
webrtc/base/testclient.h:48
↓ 3 callersMethodallowedCommandNames
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:297
↓ 3 callersMethodappendToNewConfig
AnyCore/lib_rtsp/liveMedia/MPEG4VideoStreamFramer.cpp:132
↓ 3 callersMethodassertTrue
(boolean condition)
webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java:292
↓ 3 callersMethodassertTrue
(boolean condition)
webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java:408
↓ 3 callersMethodassertTrue
(boolean condition)
Prj-Android/app/src/main/java/org/webrtc/voiceengine/WebRtcAudioManager.java:290
↓ 3 callersMethodassertTrue
(boolean condition)
Prj-Android/app/src/main/java/org/webrtc/voiceengine/WebRtcAudioEffects.java:407
↓ 3 callersMethodattrVal_unsigned
also returns 0 if attribute doesn't exist (and has no default value)
AnyCore/lib_rtsp/liveMedia/include/MediaSession.hh:224
↓ 3 callersFunctionbadAddressForUs
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:570
↓ 3 callersMethodbytes
AnyCore/srs_librtmp/srs_librtmp.cpp:19014
↓ 3 callersMethodbytesInUse
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:73
↓ 3 callersMethodbytesInUse
AnyCore/lib_rtsp/liveMedia/AVIFileSink.cpp:44
↓ 3 callersMethodc1_validate_digest
AnyCore/srs_librtmp/srs_librtmp.cpp:26801
↓ 3 callersMethodcompleted
AnyCore/srs_librtmp/srs_librtmp.cpp:15891
↓ 3 callersMethodcomputeDigestResponse
AnyCore/lib_rtsp/liveMedia/DigestAuthentication.cpp:91
↓ 3 callersMethodconnectionEndpointName
AnyCore/lib_rtsp/liveMedia/include/MediaSession.hh:70
↓ 3 callersMethodcontext
webrtc/modules/utility/include/jvm_android.h:170
↓ 3 callersMethodcontinueAfterLivenessCommand
AnyCore/lib_rtsp/liveMedia/ProxyServerMediaSession.cpp:260
↓ 3 callersMethodcreate
AnyCore/srs_librtmp/srs_librtmp.cpp:13091
↓ 3 callersMethoddata
webrtc/base/copyonwritebuffer.h:69
↓ 3 callersMethoddata
AnyCore/lib_rtsp/groupsock/include/NetAddress.hh:51
↓ 3 callersFunctiondctsub
webrtc/common_audio/fft4g.c:1292
↓ 3 callersMethoddeleteStream
AnyCore/lib_rtsp/liveMedia/ServerMediaSession.cpp:395
↓ 3 callersMethoddrawOes
Functions for drawing frames with different sources. The rendering surface target is implied by the current EGL context of the calling thread and requ
Prj-Android/app/src/main/java/org/webrtc/RendererCommon.java:43
↓ 3 callersMethoddrawRectangle
(int x, int y, int width, int height)
Prj-Android/app/src/main/java/org/webrtc/GlRectDrawer.java:164
↓ 3 callersMethodenqueue
AnyCore/lib_rtsp/liveMedia/MediaSink.cpp:132
↓ 3 callersMethodfilename
webrtc/base/pathutils.cc:218
↓ 3 callersMethodfindByCSeq
AnyCore/lib_rtsp/liveMedia/RTSPClient.cpp:1823
↓ 3 callersFunctionfloat_Sort
webrtc/system_wrappers/source/spreadsortlib/spreadsort.hpp:933
↓ 3 callersMethodfunction_name
webrtc/base/location.h:39
↓ 3 callersMethodget
Gets the pointer, creating the singleton if necessary. May return NULL if creation failed.
webrtc/base/referencecountedsingletonfactory.h:111
↓ 3 callersMethodgetAudioFormat
AnyCore/lib_rtsp/liveMedia/WAVAudioFileSource.cpp:80
↓ 3 callersMethodgetCameraIndex
(String deviceName)
Prj-Android/app/src/main/java/org/webrtc/Camera1Enumerator.java:152
↓ 3 callersMethodgetCurrentPTS
AnyCore/lib_rtsp/liveMedia/MPEG1or2VideoStreamFramer.cpp:112
↓ 3 callersMethodgetDeviceName
@deprecated Please use Camera1Enumerator.getDeviceNames().get(index) instead.
Prj-Android/app/src/main/java/org/webrtc/CameraEnumerationAndroid.java:140
↓ 3 callersMethodgetHeight
()
Prj-Android/app/src/main/java/org/webrtc/GlTextureFrameBuffer.java:103
↓ 3 callersFunctiongetLine
AnyCore/lib_rtsp/liveMedia/SIPClient.cpp:164
↓ 3 callersMethodgetNextFrame
AnyCore/lib_rtsp/liveMedia/StreamReplicator.cpp:70
↓ 3 callersMethodgetWidth
()
Prj-Android/app/src/main/java/org/webrtc/GlTextureFrameBuffer.java:99
↓ 3 callersMethodget_property
AnyCore/srs_librtmp/srs_librtmp.cpp:19509
↓ 3 callersMethodhandleCmd_bad
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:602
↓ 3 callersMethodhasHintTrack
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:127
↓ 3 callersMethodhas_ssrc
webrtc/media/base/streamparams.h:93
↓ 3 callersMethodignored
Indicates whether this network should be ignored, perhaps because the IP is 0, or the interface is one we know is invalid.
webrtc/base/network.h:362
↓ 3 callersFunctionilog
AnyCore/lib_rtsp/liveMedia/OggFileParser.cpp:291
↓ 3 callersFunctionincreaseReceiveBufferTo
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:415
↓ 3 callersFunctioninitializeWinsockIfNecessary
AnyCore/lib_rtsp/groupsock/inet.c:25
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