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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
↓ 3 callers
Method
ParseBitstream
webrtc/modules/video_coding/utility/h264_bitstream_parser.cc:277
↓ 3 callers
Method
Playing
webrtc/modules/audio_device/audio_device_impl.cc:1495
↓ 3 callers
Method
Playing
webrtc/modules/audio_device/android/audio_track_jni.h:78
↓ 3 callers
Function
PointerTojlong
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 callers
Function
PostTaskAndReply
webrtc/base/task_queue.h:195
↓ 3 callers
Function
PrintPlane
webrtc/common_video/libyuv/webrtc_libyuv.cc:93
↓ 3 callers
Method
Receive
webrtc/modules/audio_device/android/opensles_common.h:34
↓ 3 callers
Method
Recording
webrtc/modules/audio_device/audio_device_impl.cc:1524
↓ 3 callers
Method
Recording
webrtc/modules/audio_device/android/audio_record_jni.h:82
↓ 3 callers
Method
Recording
webrtc/modules/audio_device/ios/audio_device_ios.h:68
↓ 3 callers
Method
Recv
webrtc/base/asyncsocket.cc:67
↓ 3 callers
Method
RegisterNatives
webrtc/modules/utility/source/jvm_android.cc:186
↓ 3 callers
Method
RemoveConnection
webrtc/base/virtualsocketserver.cc:737
↓ 3 callers
Method
RemoveSink
webrtc/media/base/videocapturer.cc:189
↓ 3 callers
Method
ReportFramerate
Report framerate(fps) to estimate # of samples.
webrtc/modules/video_coding/utility/quality_scaler.cc:83
↓ 3 callers
Method
ReportQP
webrtc/modules/video_coding/utility/quality_scaler.cc:88
↓ 3 callers
Method
Resample10Msec
webrtc/modules/audio_coding/acm2/acm_resampler.cc:28
↓ 3 callers
Function
SHA1Transform
Hash a single 512-bit block. This is the core of the algorithm.
webrtc/base/sha1.cc:164
↓ 3 callers
Function
SHA1Update
Run your data through this.
webrtc/base/sha1.cc:233
↓ 3 callers
Method
ScaleFrom
webrtc/common_video/video_frame_buffer.cc:214
↓ 3 callers
Function
Schedule
AnyCore/lib_rtsp/liveMedia/RTCP.cpp:984
↓ 3 callers
Method
SetAllowBlockingCalls
webrtc/base/thread.cc:296
↓ 3 callers
Method
SetApplyRotation
webrtc/media/base/videoframefactory.h:50
↓ 3 callers
Method
SetFilename
webrtc/base/pathutils.cc:224
↓ 3 callers
Method
SetMessageQueue
webrtc/base/win32socketserver.cc:764
↓ 3 callers
Method
SetPlayoutSampleRate
webrtc/modules/audio_device/include/fake_audio_device.h:140
↓ 3 callers
Method
SetSpeakerVolume
webrtc/modules/audio_device/android/audio_track_jni.cc:173
↓ 3 callers
Function
SetTraceValue
std::string version of SetTraceValue so that trace arguments can be strings.
webrtc/base/trace_event.h:803
↓ 3 callers
Method
SetVideoRender
AnyCore/anyrtmplayer.cc:72
↓ 3 callers
Method
Start
webrtc/modules/utility/source/process_thread_impl.cc:61
↓ 3 callers
Method
StartRtmpPlay
(final String strUrl, final long renderPointer)
Prj-Android/app/src/main/java/org/anyrtc/core/RTMPGuestKit.java:68
↓ 3 callers
Method
StartRtmpStream
(final String strUrl)
Prj-Android/app/src/main/java/org/anyrtc/core/RTMPHosterKit.java:149
↓ 3 callers
Method
Subtract
webrtc/common_types.h:824
↓ 3 callers
Method
Terminate
webrtc/modules/audio_device/include/fake_audio_device.h:41
↓ 3 callers
Method
ToJava
Prj-Android/jni/jni_util/native_handle_impl.cc:35
↓ 3 callers
Method
ToString
webrtc/media/base/codec.cc:199
↓ 3 callers
Function
TruncateIP
webrtc/base/ipaddress.cc:337
↓ 3 callers
Function
Type2String
webrtc/base/flags.cc:63
↓ 3 callers
Function
UNLOCK_FUNCTION
webrtc/base/thread_checker.h:98
↓ 3 callers
Function
Utf8ToWindowsFilename
webrtc/base/win32.cc:369
↓ 3 callers
Method
VerifyAndAllocateFragmentationHeader
webrtc/modules/include/module_common_types.h:387
↓ 3 callers
Function
WebRtcSpl_AddSatW16
webrtc/common_audio/signal_processing/include/spl_inl.h:114
↓ 3 callers
Function
WebRtcSpl_CreateRealFFT
webrtc/common_audio/signal_processing/real_fft.c:21
↓ 3 callers
Function
WebRtcSpl_DivW32W16
webrtc/common_audio/signal_processing/division_operations.c:38
↓ 3 callers
Function
WebRtcSpl_GetSizeInBits
webrtc/common_audio/signal_processing/include/spl_inl.h:124
↓ 3 callers
Function
WebRtcSpl_ResampDotProduct
compute two inner-products and store them to output array
webrtc/common_audio/signal_processing/resample_fractional.c:147
↓ 3 callers
Function
WebRtcSpl_Resample22khzTo16khz
22 -> 16 resampler
webrtc/common_audio/signal_processing/resample.c:45
↓ 3 callers
Function
WebRtcSpl_ResetResample22khzTo16khz
initialize state of 22 -> 16 resampler
webrtc/common_audio/signal_processing/resample.c:96
↓ 3 callers
Function
WebRtcSpl_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 callers
Function
WebRtc_MoveReadPtr
webrtc/common_audio/ring_buffer.c:178
↓ 3 callers
Function
WebRtc_available_write
webrtc/common_audio/ring_buffer.c:227
↓ 3 callers
Function
_srs_expect_bandwidth_packet
AnyCore/srs_librtmp/srs_librtmp.cpp:35054
↓ 3 callers
Method
active
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 callers
Method
address
webrtc/base/testclient.h:48
↓ 3 callers
Method
allowedCommandNames
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:297
↓ 3 callers
Method
appendToNewConfig
AnyCore/lib_rtsp/liveMedia/MPEG4VideoStreamFramer.cpp:132
↓ 3 callers
Method
assertTrue
(boolean condition)
webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java:292
↓ 3 callers
Method
assertTrue
(boolean condition)
webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java:408
↓ 3 callers
Method
assertTrue
(boolean condition)
Prj-Android/app/src/main/java/org/webrtc/voiceengine/WebRtcAudioManager.java:290
↓ 3 callers
Method
assertTrue
(boolean condition)
Prj-Android/app/src/main/java/org/webrtc/voiceengine/WebRtcAudioEffects.java:407
↓ 3 callers
Method
attrVal_unsigned
also returns 0 if attribute doesn't exist (and has no default value)
AnyCore/lib_rtsp/liveMedia/include/MediaSession.hh:224
↓ 3 callers
Function
badAddressForUs
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:570
↓ 3 callers
Method
bytes
AnyCore/srs_librtmp/srs_librtmp.cpp:19014
↓ 3 callers
Method
bytesInUse
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:73
↓ 3 callers
Method
bytesInUse
AnyCore/lib_rtsp/liveMedia/AVIFileSink.cpp:44
↓ 3 callers
Method
c1_validate_digest
AnyCore/srs_librtmp/srs_librtmp.cpp:26801
↓ 3 callers
Method
completed
AnyCore/srs_librtmp/srs_librtmp.cpp:15891
↓ 3 callers
Method
computeDigestResponse
AnyCore/lib_rtsp/liveMedia/DigestAuthentication.cpp:91
↓ 3 callers
Method
connectionEndpointName
AnyCore/lib_rtsp/liveMedia/include/MediaSession.hh:70
↓ 3 callers
Method
context
webrtc/modules/utility/include/jvm_android.h:170
↓ 3 callers
Method
continueAfterLivenessCommand
AnyCore/lib_rtsp/liveMedia/ProxyServerMediaSession.cpp:260
↓ 3 callers
Method
create
AnyCore/srs_librtmp/srs_librtmp.cpp:13091
↓ 3 callers
Method
data
webrtc/base/copyonwritebuffer.h:69
↓ 3 callers
Method
data
AnyCore/lib_rtsp/groupsock/include/NetAddress.hh:51
↓ 3 callers
Function
dctsub
webrtc/common_audio/fft4g.c:1292
↓ 3 callers
Method
deleteStream
AnyCore/lib_rtsp/liveMedia/ServerMediaSession.cpp:395
↓ 3 callers
Method
drawOes
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 callers
Method
drawRectangle
(int x, int y, int width, int height)
Prj-Android/app/src/main/java/org/webrtc/GlRectDrawer.java:164
↓ 3 callers
Method
enqueue
AnyCore/lib_rtsp/liveMedia/MediaSink.cpp:132
↓ 3 callers
Method
filename
webrtc/base/pathutils.cc:218
↓ 3 callers
Method
findByCSeq
AnyCore/lib_rtsp/liveMedia/RTSPClient.cpp:1823
↓ 3 callers
Function
float_Sort
webrtc/system_wrappers/source/spreadsortlib/spreadsort.hpp:933
↓ 3 callers
Method
function_name
webrtc/base/location.h:39
↓ 3 callers
Method
get
Gets the pointer, creating the singleton if necessary. May return NULL if creation failed.
webrtc/base/referencecountedsingletonfactory.h:111
↓ 3 callers
Method
getAudioFormat
AnyCore/lib_rtsp/liveMedia/WAVAudioFileSource.cpp:80
↓ 3 callers
Method
getCameraIndex
(String deviceName)
Prj-Android/app/src/main/java/org/webrtc/Camera1Enumerator.java:152
↓ 3 callers
Method
getCurrentPTS
AnyCore/lib_rtsp/liveMedia/MPEG1or2VideoStreamFramer.cpp:112
↓ 3 callers
Method
getDeviceName
@deprecated Please use Camera1Enumerator.getDeviceNames().get(index) instead.
Prj-Android/app/src/main/java/org/webrtc/CameraEnumerationAndroid.java:140
↓ 3 callers
Method
getHeight
()
Prj-Android/app/src/main/java/org/webrtc/GlTextureFrameBuffer.java:103
↓ 3 callers
Function
getLine
AnyCore/lib_rtsp/liveMedia/SIPClient.cpp:164
↓ 3 callers
Method
getNextFrame
AnyCore/lib_rtsp/liveMedia/StreamReplicator.cpp:70
↓ 3 callers
Method
getWidth
()
Prj-Android/app/src/main/java/org/webrtc/GlTextureFrameBuffer.java:99
↓ 3 callers
Method
get_property
AnyCore/srs_librtmp/srs_librtmp.cpp:19509
↓ 3 callers
Method
handleCmd_bad
AnyCore/lib_rtsp/liveMedia/RTSPServer.cpp:602
↓ 3 callers
Method
hasHintTrack
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:127
↓ 3 callers
Method
has_ssrc
webrtc/media/base/streamparams.h:93
↓ 3 callers
Method
ignored
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 callers
Function
ilog
AnyCore/lib_rtsp/liveMedia/OggFileParser.cpp:291
↓ 3 callers
Function
increaseReceiveBufferTo
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:415
↓ 3 callers
Function
initializeWinsockIfNecessary
AnyCore/lib_rtsp/groupsock/inet.c:25
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