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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 16 callersMethodReset
modules/audio_processing/agc/agc.cc:71
↓ 16 callersFunctionSetBE16
rtc_base/byte_order.h:109
↓ 16 callersFunctionSetEstimate
modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc:63
↓ 16 callersMethodSetSend
pc/rtp_sender.cc:497
↓ 16 callersMethodShare
modules/desktop_capture/shared_desktop_frame.cc:40
↓ 16 callersFunctionTimeSince
The number of milliseconds that have elapsed since 'earlier'.
rtc_base/time_utils.h:96
↓ 16 callersMethodToI420
test/fake_texture_frame.cc:45
↓ 16 callersMethodToI420
common_video/video_frame_buffer.cc:116
↓ 16 callersFunctionVerifyPeerCertMatchesHost
rtc_base/openssl_utility.cc:68
↓ 16 callersMethodWait
call/bitrate_estimator_tests.cc:46
↓ 16 callersFunctionWebRtcOpus_DecoderFree
modules/audio_coding/codecs/opus/opus_interface.cc:468
↓ 16 callersMethodWriteSamples
modules/audio_processing/test/test_utils.cc:27
↓ 16 callersMethodaudio_frames_received
pc/peer_connection_integrationtest.cc:411
↓ 16 callersMethodbitrate
modules/congestion_controller/goog_cc/bitrate_estimator.cc:148
↓ 16 callersMethodchannel_layout
api/audio/audio_frame.h:104
↓ 16 callersMethodcreateCapturer
(String name, boolean initialize)
sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java:343
↓ 16 callersMethoddisposeCapturer
(CapturerInstance instance)
sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java:375
↓ 16 callersMethodextmap_allow_mixed
pc/session_description.h:626
↓ 16 callersMethodgetId
()
sdk/android/api/org/webrtc/RTCStats.java:49
↓ 16 callersMethodgetStrideY
()
sdk/android/api/org/webrtc/VideoFrame.java:88
↓ 16 callersMethodhas_ssrc_group
media/base/stream_params.h:122
↓ 16 callersMethodheight
api/video/i420_buffer.cc:141
↓ 16 callersMethodid
logging/rtc_event_log/events/rtc_event_probe_result_success.h:33
↓ 16 callersMethodimage
examples/peerconnection/client/main_wnd.h:123
↓ 16 callersMethodlast_recv_rtp_packet
pc/test/rtp_transport_test_util.h:50
↓ 16 callersMethodnack
test/rtcp_packet_parser.h:90
↓ 16 callersMethodreset
(int width, int height)
sdk/android/api/org/webrtc/MediaCodecVideoDecoder.java:504
↓ 16 callersMethodsecure
pc/media_session.h:167
↓ 16 callersMethodsend
test/scenario/audio_stream.h:79
↓ 16 callersMethodsize
api/array_view.h:100
↓ 16 callersMethodswapBuffers
()
sdk/android/api/org/webrtc/EglBase.java:254
↓ 16 callersMethodwidth
api/video/i420_buffer.cc:137
↓ 15 callersMethodAddMediaReceiverRtcpObserver
modules/rtp_rtcp/source/rtcp_transceiver.cc:52
↓ 15 callersFunctionAddOrUpdateSink
Register a video sink for this track. Used to connect the track to the underlying video engine.
api/media_stream_interface.h:185
↓ 15 callersFunctionBuildAv1Frame
modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc:137
↓ 15 callersMethodBuildFlexfecPacket
modules/rtp_rtcp/source/fec_test_helper.cc:159
↓ 15 callersMethodClamped
rtc_base/units/unit_base.h:226
↓ 15 callersMethodColorSpace
api/video/color_space.cc:67
↓ 15 callersMethodConnect
rtc_base/sigslot_unittest.cc:278
↓ 15 callersMethodCopyBytes
api/transport/stun.cc:950
↓ 15 callersFunctionCreateFirFilter
common_audio/fir_filter_factory.cc:26
↓ 15 callersMethodCreatePeerConnection
pc/peer_connection_factory.cc:229
↓ 15 callersFunctionCreateSpecificSimulcastTestFixture
media/engine/simulcast_encoder_adapter_unittest.cc:51
↓ 15 callersMethodCreateTransformationMatrix
audio/utility/channel_mixing_matrix.cc:97
↓ 15 callersFunctionDecode
modules/audio_coding/codecs/isac/main/source/isac.c:959
↓ 15 callersMethodDurationsMs
video/quality_limitation_reason_tracker.cc:42
↓ 15 callersMethodExtractData
test/pc/e2e/analyzer/video/default_encoded_image_data_injector.cc:63
↓ 15 callersMethodFecAvailable
modules/rtp_rtcp/source/flexfec_sender.cc:120
↓ 15 callersFunctionFileExists
test/testsupport/file_utils.cc:83
↓ 15 callersMethodFirstPacketInSubFrame
modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h:36
↓ 15 callersFunctionFromString
rtc_base/string_encode.h:131
↓ 15 callersMethodGetAverageRoundedDown
rtc_base/numerics/moving_average.cc:34
↓ 15 callersFunctionGetBoolFromJson
rtc_base/strings/json.cc:79
↓ 15 callersMethodGetConfidenceInterval
rtc_base/numerics/event_based_exponential_moving_average.cc:77
↓ 15 callersMethodGetId
Return kInvalidId if not found.
modules/rtp_rtcp/include/rtp_header_extension_map.h:48
↓ 15 callersMethodGetMethodId
JavaClass implementation.
modules/utility/source/jvm_android.cc:157
↓ 15 callersMethodGetNetworks
rtc_base/network_unittest.cc:105
↓ 15 callersMethodGetNumCreatedSendStreams
media/engine/fake_webrtc_call.cc:609
↓ 15 callersMethodGetStats
call/call.cc:1069
↓ 15 callersFunctionGetValueFromString
pc/webrtc_sdp.cc:639
↓ 15 callersMethodInitRTPVideoHeaderVP9
modules/video_coding/codecs/vp9/include/vp9_globals.h:159
↓ 15 callersMethodIsBlocking
rtc_base/socket.h:106
↓ 15 callersMethodLookup
Implements the Var syntax.
PRESUBMIT.py:1101
↓ 15 callersMethodOnReadyToSend
media/engine/webrtc_voice_engine.cc:2149
↓ 15 callersMethodOnSuspendChange
video/send_statistics_proxy.cc:697
↓ 15 callersFunctionParseLocalDepsFile
(filename)
tools_webrtc/autoroller/roll_deps.py:114
↓ 15 callersMethodPresubmitLocalPath
(self)
presubmit_test_mocks.py:40
↓ 15 callersMethodProcess
modules/audio_processing/agc/agc.cc:38
↓ 15 callersMethodPush
Replace the least recently pushed array in the buffer with |new_values|.
modules/audio_processing/agc2/rnn_vad/ring_buffer.h:39
↓ 15 callersFunctionQ32x32ToInt64Ms
Converts Q32.32-formatted fixed-point seconds to |int64_t| milliseconds.
system_wrappers/include/ntp_time.h:104
↓ 15 callersMethodReadUInt32
rtc_base/bit_buffer.cc:106
↓ 15 callersMethodReconfigureVideoEncoder
video/video_send_stream.cc:172
↓ 15 callersMethodRecv
rtc_base/async_socket.cc:67
↓ 15 callersMethodRemoveStream
pc/peer_connection.cc:1481
↓ 15 callersFunctionRunTest
modules/audio_processing/rms_level_unittest.cc:28
↓ 15 callersMethodSSRC
modules/rtp_rtcp/source/rtp_sender.h:126
↓ 15 callersMethodSetJitter
modules/rtp_rtcp/source/rtcp_packet/report_block.h:44
↓ 15 callersMethodSetPacketIds
modules/rtp_rtcp/source/rtcp_packet/nack.cc:127
↓ 15 callersMethodSetPayloadType
video/video_receive_stream_unittest.cc:75
↓ 15 callersMethodSetPeerConnectionInterface
pc/test/mock_peer_connection_observers.h:61
↓ 15 callersMethodSetProvisionalAnswer
pc/srtp_filter.cc:71
↓ 15 callersMethodSetRates
media/engine/simulcast_encoder_adapter_unittest.cc:228
↓ 15 callersMethodSetRecordingSampleRate
modules/audio_device/win/audio_device_module_win.cc:481
↓ 15 callersMethodSetRemb
modules/rtp_rtcp/source/rtcp_sender.cc:269
↓ 15 callersMethodSetRemoteSSRC
modules/rtp_rtcp/source/rtcp_sender.cc:334
↓ 15 callersMethodSetRtcpPacketTransport
pc/rtp_transport.cc:86
↓ 15 callersMethodSetWindowSize
Set the window size over which to measure the current bitrate. For retransmissions, this is typically the RTT.
rtc_base/rate_limiter.cc:62
↓ 15 callersMethodToInternal
modules/audio_coding/neteq/timestamp_scaler.cc:23
↓ 15 callersFunctionToRtcpFeedback
pc/rtp_parameters_conversion.cc:222
↓ 15 callersMethodTransform
audio/utility/channel_mixer.cc:34
↓ 15 callersFunctionTruncateIP
rtc_base/ip_address.cc:357
↓ 15 callersMethodUpdate
modules/remote_bitrate_estimator/overuse_estimator.cc:48
↓ 15 callersMethodUpdateFrame
api/audio/audio_frame.cc:46
↓ 15 callersFunctionVP8Get
modules/video_coding/utility/vp8_header_parser.cc:110
↓ 15 callersFunctionWebRtcIsacfix_EncHistMulti
* WebRtcIsacfix_EncHistMulti(...) * * Encode the histogram interval * * Input: * - streamData : in-/output struct containing bitstrea
modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c:35
↓ 15 callersFunctionWebRtcIsacfix_Log2Q8
modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c:25
↓ 15 callersFunctionWebRtcSpl_DivW32W16ResW16
common_audio/signal_processing/division_operations.c:51
↓ 15 callersMethodWriteFrame
test/testsupport/yuv_frame_writer.cc:53
↓ 15 callersMethodbegin
rtc_tools/rtc_event_log_visualizer/main.cc:163
↓ 15 callersMethoddata
media/sctp/sctp_transport.h:122
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