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

⨍ Functions 10,022◇ Types & classes 2,276
↓ 2 callersMethodInitDecode
webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc:192
↓ 2 callersMethodInitEncode
Prj-Android/jni/androidmediaencoder_jni.cc:366
↓ 2 callersMethodInitPlayout
webrtc/modules/audio_device/android/audio_track_jni.cc:46
↓ 2 callersMethodInitialize
webrtc/modules/audio_device/win/audio_device_core_win.h:71
↓ 2 callersMethodInitialized
webrtc/modules/audio_device/android/audio_device_template.h:86
↓ 2 callersMethodInitialized
webrtc/modules/audio_device/ios/audio_device_ios.h:54
↓ 2 callersFunctionInterlockedExchangePointer
The problem is that the macro provided for InterlockedExchangePointer() is doing a (LONG) C-style cast that triggers invariably the warning C4312 when
webrtc/system_wrappers/include/fix_interlocked_exchange_pointer_win.h:30
↓ 2 callersFunctionInternalUrlDecodeString
webrtc/base/urlencode.cc:136
↓ 2 callersMethodIsAcousticEchoCancelerSupported
webrtc/modules/audio_device/android/audio_manager.cc:185
↓ 2 callersMethodIsAutomaticGainControlSupported
webrtc/modules/audio_device/android/audio_manager.cc:190
↓ 2 callersMethodIsCommunicationModeEnabled
webrtc/modules/audio_device/android/audio_manager.cc:58
↓ 2 callersMethodIsConnected
webrtc/base/testutils.h:316
↓ 2 callersMethodIsDirectory
returns true if the file currently pointed to is a directory
webrtc/base/fileutils.cc:102
↓ 2 callersMethodIsEmpty
AnyCore/lib_rtsp/groupsock/include/NetInterface.hh:61
↓ 2 callersFunctionIsH264CodecSupported
If any H.264 codec is supported (iOS HW or OpenH264/FFmpeg).
webrtc/modules/video_coding/codecs/h264/h264.cc:50
↓ 2 callersFunctionIsHostBigEndian
Check if the current host is big endian.
webrtc/base/byteorder.h:129
↓ 2 callersFunctionIsIgnoredIPv6
webrtc/base/network.cc:119
↓ 2 callersFunctionIsNewerSequenceNumber
webrtc/modules/include/module_common_types.h:699
↓ 2 callersMethodIsNoiseSuppressorSupported
webrtc/modules/audio_device/android/audio_manager.cc:195
↓ 2 callersMethodIsPlaying
AnyCore/plydecoder.cc:251
↓ 2 callersMethodIsValid
webrtc/base/window.h:62
↓ 2 callersFunctionKeepRefUntilDone
webrtc/base/keep_ref_until_done.h:30
↓ 2 callersMethodLock
webrtc/modules/audio_device/win/audio_device_wave_win.h:167
↓ 2 callersMethodLockShared
webrtc/base/sharedexclusivelock.cc:29
↓ 2 callersFunctionLogAssert
webrtc/base/common.cc:54
↓ 2 callersFunctionLookUpClass
webrtc/modules/utility/source/jvm_android.cc:61
↓ 2 callersMethodLookupConnection
webrtc/base/virtualsocketserver.cc:725
↓ 2 callersFunctionMP3HuffmanDecode
AnyCore/lib_rtsp/liveMedia/MP3InternalsHuffman.cpp:541
↓ 2 callersMethodMatches
webrtc/media/base/streamparams.h:172
↓ 2 callersMethodMaxSpeakerVolume
webrtc/modules/audio_device/android/audio_track_jni.cc:179
↓ 2 callersMethodMaybeRenderDecodedTextureBuffer
()
Prj-Android/app/src/main/java/org/webrtc/MediaCodecVideoDecoder.java:689
↓ 2 callersMethodMinSample
webrtc/system_wrappers/source/metrics_default.cc:86
↓ 2 callersMethodMoveReadPositionBackward
webrtc/common_audio/audio_ring_buffer.cc:67
↓ 2 callersMethodMuted
AnyCore/avcodec.h:55
↓ 2 callersMethodNewMicLevel
webrtc/modules/audio_device/audio_device_buffer.cc:276
↓ 2 callersFunctionNewSender
AnyCore/lib_rtsp/liveMedia/RTCP.cpp:1047
↓ 2 callersMethodNumEvents
webrtc/system_wrappers/source/metrics_default.cc:71
↓ 2 callersMethodNumSamples
webrtc/system_wrappers/source/metrics_default.cc:77
↓ 2 callersMethodNumberOfDevices
webrtc/modules/video_capture/windows/device_info_mf.cc:26
↓ 2 callersMethodOnEncodeDataCallback
AnyCore/anyrtmpstreamer.cc:144
↓ 2 callersMethodOnMovedFrom
Called when *this has been moved from. Conceptually it's a no-op, but we can mutate the state slightly to help subsequent sanity checks catch bugs.
webrtc/base/buffer.h:335
↓ 2 callersMethodOnRecordAudio
For webrtc::AVAudioRecordCallback
AnyCore/RtmpHosterImpl.cc:313
↓ 2 callersMethodOnRtmplayerCache
(int time)
Prj-Android/app/src/main/java/org/anyrtc/core/RTMPGuestHelper.java:28
↓ 2 callersMethodOnRtmplayerOK
()
Prj-Android/app/src/main/java/org/anyrtc/core/RTMPGuestHelper.java:26
↓ 2 callersMethodOnRtmplayerStatus
(int cacheTime, int curBitrate)
Prj-Android/app/src/main/java/org/anyrtc/core/RTMPGuestHelper.java:27
↓ 2 callersMethodOpenFile
webrtc/base/unixfilesystem.cc:125
↓ 2 callersMethodOpenMicrophone
webrtc/modules/audio_device/win/audio_mixer_manager_win.cc:794
↓ 2 callersMethodOpenSpeaker
webrtc/modules/audio_device/win/audio_mixer_manager_win.cc:572
↓ 2 callersFunctionParseNaluType
webrtc/common_video/h264/h264_common.cc:57
↓ 2 callersMethodPause
webrtc/modules/video_capture/windows/sink_filter_ds.cc:424
↓ 2 callersFunctionPerformLogging
webrtc/system_wrappers/source/data_log_unittest_disabled.cc:21
↓ 2 callersMethodPing
webrtc/base/winping.cc:211
↓ 2 callersMethodPlayoutIsInitialized
webrtc/modules/audio_device/audio_device_impl.cc:1455
↓ 2 callersFunctionPrintFlagValue
webrtc/base/flags.cc:74
↓ 2 callersMethodPush
Synchronous resampling, all output samples are written to samplesOut
webrtc/common_audio/resampler/resampler.cc:404
↓ 2 callersFunctionPutMP3SideInfoIntoFrame
AnyCore/lib_rtsp/liveMedia/MP3Internals.cpp:620
↓ 2 callersMethodRTPgs
AnyCore/lib_rtsp/liveMedia/include/RTPSource.hh:44
↓ 2 callersMethodRate
webrtc/base/rate_statistics.cc:64
↓ 2 callersMethodRead
webrtc/base/stream.cc:166
↓ 2 callersMethodReadSignedExponentialGolomb
webrtc/base/bitbuffer.cc:190
↓ 2 callersFunctionReceivedPacketSize
AnyCore/lib_rtsp/liveMedia/RTCP.cpp:1033
↓ 2 callersMethodRecordingIsInitialized
webrtc/modules/audio_device/android/audio_record_jni.h:78
↓ 2 callersMethodRecordingIsInitialized
webrtc/modules/audio_device/ios/audio_device_ios.h:60
↓ 2 callersMethodRegisterAudioCallback
webrtc/modules/audio_device/audio_device_impl.cc:1547
↓ 2 callersFunctionRemoveFirst
webrtc/media/base/streamparams.cc:212
↓ 2 callersFunctionReplySize
webrtc/base/winping.cc:129
↓ 2 callersMethodReset
Functions only for testing.
webrtc/system_wrappers/source/metrics_default.cc:66
↓ 2 callersMethodResetIfNeeded
webrtc/common_audio/resampler/resampler.cc:79
↓ 2 callersMethodReturnTextureFrame
Prj-Android/jni/surfacetexturehelper_jni.cc:66
↓ 2 callersFunctionRiffChunkSize
webrtc/common_audio/wav_header.cc:133
↓ 2 callersMethodRotate
Prj-Android/jni/jni_util/native_handle_impl.cc:41
↓ 2 callersMethodRun
static
webrtc/modules/utility/source/process_thread_impl.cc:187
↓ 2 callersMethodRun
webrtc/base/platform_thread.cc:191
↓ 2 callersMethodRun
webrtc/common_audio/resampler/push_sinc_resampler.cc:81
↓ 2 callersFunctionSHAPrintContext
SAK
webrtc/base/sha1.cc:151
↓ 2 callersMethodSet
webrtc/system_wrappers/source/event_timer_win.cc:35
↓ 2 callersMethodSet
TODO(pbos): Make this void.
webrtc/system_wrappers/source/event_timer_posix.cc:66
↓ 2 callersMethodSetActiveAudioLayer
A valid audio layer must always be set before calling Init(), hence we might as well make it a part of the test fixture.
webrtc/modules/audio_device/android/audio_manager_unittest.cc:43
↓ 2 callersMethodSetBasename
webrtc/base/pathutils.cc:193
↓ 2 callersFunctionSetClockForTesting
webrtc/base/timeutils.cc:35
↓ 2 callersMethodSetCurrentMicLevel
webrtc/modules/audio_device/audio_device_buffer.cc:260
↓ 2 callersMethodSetData
webrtc/base/stream.cc:640
↓ 2 callersMethodSetData
AnyCore/plybuffer.h:36
↓ 2 callersMethodSetError
webrtc/base/asynctcpsocket.cc:129
↓ 2 callersMethodSetIP
webrtc/base/socketaddress.cc:89
↓ 2 callersMethodSetIPs
Sets the addresses of this network. Returns true if the address set changed. Change detection is short circuited if the changed argument is true.
webrtc/base/network.cc:923
↓ 2 callersMethodSetLocalAddress
webrtc/base/virtualsocketserver.cc:136
↓ 2 callersFunctionSetNonBlocking
webrtc/base/task_queue_libevent.cc:39
↓ 2 callersMethodSetPlayoutSampleRate
TODO(henrika): remove
webrtc/modules/audio_device/android/audio_device_template.h:476
↓ 2 callersMethodSetPort
webrtc/base/socketaddress.cc:122
↓ 2 callersMethodSetRates
AnyCore/avcodec.cc:227
↓ 2 callersMethodSetResolvedIP
webrtc/base/socketaddress.cc:112
↓ 2 callersMethodSetTypingStatus
webrtc/modules/audio_device/audio_device_buffer.cc:266
↓ 2 callersFunctionSetupEventTracer
webrtc/base/event_tracer.cc:33
↓ 2 callersFunctionSincScaleFactor
webrtc/common_audio/resampler/sinc_resampler.cc:103
↓ 2 callersMethodSleepCS
webrtc/system_wrappers/source/condition_variable_event_win.cc:121
↓ 2 callersFunctionSsrcsToString
webrtc/media/base/streamparams.cc:86
↓ 2 callersMethodStart
webrtc/base/event_tracer.cc:141
↓ 2 callersMethodStartPlayout
webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc:567
↓ 2 callersMethodStop
webrtc/modules/video_capture/windows/sink_filter_ds.cc:450
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