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

⨍ Functions 10,022◇ Types & classes 2,276
↓ 3 callersFunctiontimevalGE
AnyCore/lib_rtsp/liveMedia/QuickTimeFileSink.cpp:468
↓ 3 callersMethodto_boolean
AnyCore/srs_librtmp/srs_librtmp.cpp:19175
↓ 3 callersMethodto_ecma_array
AnyCore/srs_librtmp/srs_librtmp.cpp:19210
↓ 3 callersMethodtotalBytesAvailable
AnyCore/lib_rtsp/liveMedia/include/MediaSink.hh:82
↓ 3 callersMethodtrackId
AnyCore/lib_rtsp/liveMedia/ServerMediaSession.cpp:352
↓ 3 callersFunctiontransform
webrtc/base/stringencode.cc:540
↓ 3 callersMethodtransportPacketNumber
AnyCore/lib_rtsp/liveMedia/MPEG2IndexFromTransportStream.cpp:57
↓ 3 callersFunctionuLawFrom16BitLinear
AnyCore/lib_rtsp/liveMedia/uLawAudioFilter.cpp:74
↓ 3 callersMethodunsetSpecificRRHandler
AnyCore/lib_rtsp/liveMedia/RTCP.cpp:275
↓ 3 callersMethodupdateStateOnPlayChange
AnyCore/lib_rtsp/liveMedia/MPEG2TransportFileServerMediaSubsession.cpp:318
↓ 3 callersMethoduseProgram
()
Prj-Android/app/src/main/java/org/webrtc/GlShader.java:110
↓ 3 callersMethodv4AddressAsHostOrderInteger
webrtc/base/ipaddress.cc:49
↓ 3 callersFunctionvalidateCommentHeader
AnyCore/lib_rtsp/liveMedia/OggFileParser.cpp:640
↓ 3 callersFunctionvsprintfn
webrtc/base/stringutils.h:209
↓ 3 callersFunctionwriteSocket
AnyCore/lib_rtsp/groupsock/GroupsockHelper.cpp:315
↓ 3 callersMethodwrite_header
AnyCore/srs_librtmp/srs_librtmp.cpp:13255
↓ 3 callersMethodwrite_string
AnyCore/srs_librtmp/srs_librtmp.cpp:11913
↓ 2 callersMethodActiveAudioLayer
webrtc/modules/audio_device/android/audio_device_template.h:49
↓ 2 callersMethodAdaptFrame
webrtc/media/base/videocapturer.cc:217
↓ 2 callersFunctionAdd
NOLINT
webrtc/base/mod_ops.h:22
↓ 2 callersFunctionAddBitstreamRestriction
Adds a bitstream restriction VUI segment.
webrtc/common_video/h264/sps_vui_rewriter.cc:316
↓ 2 callersMethodAddFrame
webrtc/common_video/video_render_frames.cc:36
↓ 2 callersFunctionAddMember
AnyCore/lib_rtsp/liveMedia/RTCP.cpp:1051
↓ 2 callersMethodAddRef
webrtc/modules/audio_device/include/fake_audio_device.h:19
↓ 2 callersFunctionAlignedMalloc
webrtc/system_wrappers/include/aligned_malloc.h:45
↓ 2 callersFunctionApplyWindow
Pointwise multiplies each channel of |frames| with |window|. Results are stored in |frames|.
webrtc/common_audio/blocker.cc:76
↓ 2 callersMethodArray
webrtc/system_wrappers/include/aligned_array.h:42
↓ 2 callersMethodAtEnd
Prj-Android/jni/jni_util/jni_helpers.cc:345
↓ 2 callersFunctionBlockAlign
webrtc/common_audio/wav_header.cc:143
↓ 2 callersFunctionByteRate
webrtc/common_audio/wav_header.cc:138
↓ 2 callersMethodCacheTime
AnyCore/plydecoder.cc:262
↓ 2 callersMethodClose
VideoRender/pipeline_render.cpp:404
↓ 2 callersMethodClose
Prj-Android/jni/jRTMPGuestImpl.cc:43
↓ 2 callersFunctionCloseOutputFile
AnyCore/lib_rtsp/liveMedia/OutputFile.cpp:57
↓ 2 callersMethodCloseSpeaker
webrtc/modules/audio_device/win/audio_mixer_manager_win.cc:86
↓ 2 callersFunctionCompensateForWrapAround
Detects if there has been a wraparound between |old_timestamp| and |new_timestamp|, and compensates by adding 2^32 if that is the case.
webrtc/system_wrappers/source/rtp_to_ntp.cc:44
↓ 2 callersFunctionConjugate
webrtc/common_audio/real_fourier_ooura.cc:25
↓ 2 callersMethodConsumeBytes
webrtc/base/bitbuffer.cc:144
↓ 2 callersFunctionConvertVideoType
webrtc/common_video/libyuv/webrtc_libyuv.cc:205
↓ 2 callersFunctionCopyHrdParameters
Copies a VUI HRD parameters segment.
webrtc/common_video/h264/sps_vui_rewriter.cc:283
↓ 2 callersMethodCreate
webrtc/base/physicalsocketserver.cc:152
↓ 2 callersMethodCreate
Creates an instance of DBusMonitor
webrtc/base/dbus.cc:316
↓ 2 callersMethodCreate
Prj-Win/VideoRendererWin32.cpp:25
↓ 2 callersFunctionCreateCFDictionary
Convenience function for creating a dictionary.
webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc:33
↓ 2 callersFunctionCreatePacketTime
webrtc/base/asyncpacketsocket.h:62
↓ 2 callersMethodCrop
Center crop by keeping xFraction of the width and yFraction of the height, so e.g. cropping from 640x480 to 640x360 would use xFraction=1, yFraction=3
Prj-Android/jni/jni_util/native_handle_impl.cc:94
↓ 2 callersMethodCurrentNtp
webrtc/system_wrappers/source/clock.cc:251
↓ 2 callersMethodDeRegisterCaptureDataCallback
webrtc/modules/video_capture/video_capture_impl.cc:184
↓ 2 callersMethodDestroyDeviceInfo
webrtc/media/engine/webrtcvideocapturer.cc:59
↓ 2 callersFunctionEmptySocketAddressWithFamily
webrtc/base/socketaddress.cc:330
↓ 2 callersMethodEquals
webrtc/base/window.h:63
↓ 2 callersMethodExecutor
()
Prj-Android/app/src/main/java/org/anyrtc/core/AnyRTMP.java:65
↓ 2 callersFunctionFileIsSeekable
AnyCore/lib_rtsp/liveMedia/InputFile.cpp:105
↓ 2 callersFunctionFileOpen
webrtc/system_wrappers/source/file_impl.cc:24
↓ 2 callersFunctionFindClass
webrtc/modules/utility/source/helpers_android.cc:66
↓ 2 callersFunctionFindNaluIndices
webrtc/common_video/h264/h264_common.cc:18
↓ 2 callersFunctionFinishPath
webrtc/base/pathutils.h:130
↓ 2 callersFunctionFloatS16ToS16
webrtc/common_audio/audio_util.cc:27
↓ 2 callersMethodFlush
webrtc/system_wrappers/source/data_log.cc:214
↓ 2 callersMethodFormat
webrtc/base/window.h:51
↓ 2 callersMethodFrameToRender
webrtc/common_video/video_render_frames.cc:63
↓ 2 callersFunctionGenerateFakeSps
Generates a fake SPS with basically everything empty but the width/height. Pass in a buffer of at least kSpsBufferMaxSize. The fake SPS that this gene
webrtc/common_video/h264/sps_parser_unittest.cc:44
↓ 2 callersFunctionGetADUInfoFromMP3Frame
AnyCore/lib_rtsp/liveMedia/MP3Internals.cpp:303
↓ 2 callersMethodGetAndReset
Returns a copy (or nullptr if there are no samples) and clears samples.
webrtc/system_wrappers/source/metrics_default.cc:51
↓ 2 callersMethodGetAndroidAudioManagerForTest
Only use this acccessor for test purposes on Android.
webrtc/modules/audio_device/audio_device_impl.h:195
↓ 2 callersMethodGetAudioDeviceBuffer
webrtc/modules/audio_device/audio_device_impl.h:199
↓ 2 callersFunctionGetBE16
webrtc/base/byteorder.h:61
↓ 2 callersMethodGetBytesPer10msBuffer
webrtc/modules/audio_device/include/audio_device_defines.h:187
↓ 2 callersMethodGetDeviceModel
webrtc/modules/audio_device/android/build_info.cc:29
↓ 2 callersMethodGetDeviceName
webrtc/modules/video_capture/windows/device_info_mf.cc:30
↓ 2 callersFunctionGetDllError
webrtc/base/latebindingsymboltable.cc:27
↓ 2 callersMethodGetError
webrtc/base/asyncudpsocket.cc:93
↓ 2 callersFunctionGetInputPin
webrtc/modules/video_capture/windows/help_functions_ds.cc:35
↓ 2 callersFunctionGetMethodID
webrtc/modules/utility/source/helpers_android.cc:48
↓ 2 callersFunctionGetNextCallbackTime
webrtc/modules/utility/source/process_thread_impl.cc:27
↓ 2 callersMethodGetOption
webrtc/base/virtualsocketserver.cc:372
↓ 2 callersMethodGetOrCreateEntry
webrtc/base/diskcache.cc:301
↓ 2 callersFunctionGetQueuePtrTls
webrtc/base/task_queue_posix.cc:24
↓ 2 callersMethodGetRenderPointer
()
Prj-Android/app/src/main/java/org/webrtc/VideoRenderer.java:132
↓ 2 callersFunctionGetSLErrorString
Returns a string representation given an integer SL_RESULT_XXX code. The mapping can be found in <SLES/OpenSLES.h>.
webrtc/modules/audio_device/android/opensles_common.cc:25
↓ 2 callersFunctionGetSocketRecvTimestamp
webrtc/base/physicalsocketserver.cc:64
↓ 2 callersFunctionGetStream
webrtc/media/base/streamparams.h:234
↓ 2 callersMethodGetSupportedFormats
webrtc/media/base/videocapturer.cc:82
↓ 2 callersMethodGetTargetBitrateBps
webrtc/common_video/bitrate_adjuster.cc:63
↓ 2 callersMethodGetVideoSink
AnyCore/anyrtmpstreamer.h:41
↓ 2 callersMethodGotSelfPtr
AnyCore/RtmpHosterImpl.h:50
↓ 2 callersMethodGuest
Prj-Android/jni/jRTMPGuestImpl.h:31
↓ 2 callersMethodHas
webrtc/media/base/codec.cc:34
↓ 2 callersFunctionHighestBits
Returns the highest (left-most) |bit_count| bits in |byte|, shifted to the lowest bits (to the right).
webrtc/base/bitbuffer.cc:28
↓ 2 callersFunctionI420DataSize
webrtc/common_video/video_frame_buffer.cc:27
↓ 2 callersFunctionIPIsLinkLocal
webrtc/base/ipaddress.cc:443
↓ 2 callersFunctionIPIsULA
webrtc/base/ipaddress.cc:464
↓ 2 callersFunctionIPIsV4Mapped
webrtc/base/ipaddress.cc:478
↓ 2 callersFunctionIncreaseSeed
webrtc/common_audio/signal_processing/randomization_functions.c:93
↓ 2 callersMethodInit
webrtc/media/engine/webrtcvideoframe.cc:46
↓ 2 callersMethodInit
webrtc/modules/audio_device/android/audio_manager.cc:50
↓ 2 callersMethodInit
webrtc/modules/audio_device/include/fake_audio_device.h:27
↓ 2 callersMethodInit
webrtc/modules/video_coding/utility/quality_scaler.cc:46
↓ 2 callersMethodInit
(final Context ctx)
Prj-Android/app/src/main/java/org/anyrtc/core/AnyRTMP.java:56
↓ 2 callersMethodInit
Prj-Android/jni/video_render.cc:46
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