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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 39 callersMethodFilter
common_audio/fir_filter_c.cc:34
↓ 39 callersMethodInsertDecoded
modules/video_coding/utility/decoded_frames_history.cc:29
↓ 39 callersMethodQueueSizePackets
modules/pacing/pacing_controller.cc:250
↓ 39 callersMethodSetState
Set state of frame
modules/video_coding/frame_buffer.cc:227
↓ 39 callersFunctionToHex
rtc_base/string_utils.cc:46
↓ 39 callersFunctionToQueuedTask
rtc_base/task_utils/to_queued_task.h:58
↓ 39 callersMethodUpdateHistograms
video/call_stats.cc:198
↓ 39 callersMethodWrite
api/transport/stun.cc:491
↓ 39 callersMethodbitrate_bps
modules/rtp_rtcp/source/rtcp_packet/remb.h:37
↓ 39 callersMethodchannels
modules/audio_device/include/audio_device_defines.h:106
↓ 39 callersMethodend
rtc_tools/rtc_event_log_visualizer/main.cc:164
↓ 39 callersMethodkind
pc/audio_track.cc:41
↓ 39 callersMethodrouter
call/rtp_video_sender_unittest.cc:164
↓ 39 callersMethodxr
test/rtcp_packet_parser.h:88
↓ 38 callersMethodAddLayer
pc/simulcast_description.cc:28
↓ 38 callersMethodAddSendStream
media/engine/webrtc_video_engine_unittest.cc:2440
↓ 38 callersMethodAdvanceTimeMilliseconds
modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc:201
↓ 38 callersFunctionCreateVideoStreams
static
test/encoder_settings.cc:31
↓ 38 callersMethodExpectBidirectionalAudioAndVideo
pc/peer_connection_integrationtest.cc:1090
↓ 38 callersMethodGetAudioTracks
pc/media_stream.h:39
↓ 38 callersFunctionGetExperimentalMinVideoBitrate
rtc_base/experiments/min_video_bitrate_experiment.cc:60
↓ 38 callersMethodOnBitrateAllocationUpdated
video/send_statistics_proxy.cc:1129
↓ 38 callersFunctionParsePacket
modules/rtp_rtcp/source/flexfec_receiver_unittest.cc:39
↓ 38 callersMethodPostTask
test/pc/e2e/peer_connection_quality_test.cc:178
↓ 38 callersMethodReadUInt16
rtc_base/bit_buffer.cc:96
↓ 38 callersMethodSampleRateHz
modules/audio_coding/codecs/cng/audio_encoder_cng.cc:92
↓ 38 callersMethodSetParameters
pc/rtp_sender.cc:185
↓ 38 callersMethodSetStats
pc/test/fake_peer_connection_for_stats.h:34
↓ 38 callersFunctionWebRtcSpl_DotProductWithScale
common_audio/signal_processing/dot_product_with_scale.cc:15
↓ 38 callersMethodawait
()
sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java:615
↓ 38 callersMethoddata
modules/desktop_capture/linux/x_window_property.h:53
↓ 38 callersMethoddisplay
modules/desktop_capture/linux/x_atom_cache.cc:23
↓ 38 callersMethodget
rtc_base/thread_unittest.cc:167
↓ 38 callersMethodgetHeight
()
sdk/android/api/org/webrtc/VideoFrame.java:41
↓ 38 callersMethodgetWidth
Resolution of the buffer in pixels.
sdk/android/api/org/webrtc/VideoFrame.java:40
↓ 38 callersMethodlast_wants
video/video_stream_encoder_unittest.cc:285
↓ 38 callersMethodmedia_engine
pc/peer_connection_media_unittest.cc:51
↓ 38 callersMethodmedia_ssrc
modules/rtp_rtcp/source/rtcp_packet/psfb.h:35
↓ 38 callersMethodnegotiation_needed
pc/test/mock_peer_connection_observers.h:217
↓ 38 callersMethodpassword
p2p/base/port_unittest.cc:409
↓ 38 callersMethodsession_version
api/jsep_session_description.h:59
↓ 38 callersMethodtoString
()
sdk/android/api/org/webrtc/RTCStats.java:70
↓ 37 callersMethodAddCandidate
pc/jsep_session_description.cc:218
↓ 37 callersFunctionAttachSenderToMediaDescriptionOptions
pc/media_session_unittest.cc:345
↓ 37 callersMethodCreateDataChannel
pc/peer_connection_integrationtest.cc:393
↓ 37 callersMethodData
api/rtp_packet_infos.h:105
↓ 37 callersMethodGetOptional
rtc_base/experiments/field_trial_parser.h:191
↓ 37 callersFunctionGetValueFromJsonObject
rtc_base/strings/json.cc:242
↓ 37 callersMethodIsSending
media/engine/fake_webrtc_call.cc:149
↓ 37 callersFunctionLog
modules/congestion_controller/goog_cc/test/goog_cc_printer.cc:60
↓ 37 callersMethodOnReceiveBitrateChanged
modules/pacing/packet_router.cc:208
↓ 37 callersMethodReceivedPingResponse
p2p/base/connection.cc:881
↓ 37 callersMethodSendPacket
modules/rtp_rtcp/source/rtp_sender_unittest.cc:316
↓ 37 callersMethodSetSsrc
pc/rtp_sender.cc:256
↓ 37 callersMethodSetTransactionID
api/transport/stun.cc:83
↓ 37 callersMethodTimeNanos
rtc_base/fake_clock.cc:18
↓ 37 callersMethodToPEMString
rtc_base/fake_ssl_identity.cc:35
↓ 37 callersFunctionToString
api/candidate.h:158
↓ 37 callersMethodToString
media/base/codec.cc:235
↓ 37 callersMethodUpdate
rtc_base/openssl_digest.cc:40
↓ 37 callersFunctionWebRtcSpl_SatW32ToW16
common_audio/signal_processing/include/spl_inl.h:74
↓ 37 callersMethoddata
rtc_base/buffer_unittest.cc:101
↓ 37 callersMethodis_empty
modules/desktop_capture/desktop_geometry.h:60
↓ 37 callersMethodnum_frames_per_band
modules/audio_processing/audio_buffer.h:68
↓ 37 callersMethodport
p2p/client/basic_port_allocator.h:178
↓ 37 callersMethodremote_streams
pc/peer_connection.cc:1437
↓ 37 callersMethodverify
Implementations of verify allow applications to provide custom logic for verifying certificates. This is not required by default and should be used wi
sdk/android/api/org/webrtc/SSLCertificateVerifier.java:26
↓ 36 callersMethodAddRecvStream
media/engine/webrtc_video_engine_unittest.cc:2457
↓ 36 callersFunctionChannelLayoutToChannelCount
api/audio/channel_layout.cc:172
↓ 36 callersFunctionClipNameToClipPath
video/pc_full_stack_tests.cc:102
↓ 36 callersMethodCopy
Makes a copy os a signal. Args: signal: AudioSegment instance. Returns: An AudioSegment instance.
modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py:274
↓ 36 callersMethodGetVideoTracks
pc/media_stream.h:40
↓ 36 callersMethodMutableDataY
api/video/i010_buffer.cc:189
↓ 36 callersMethodOnSinkWants
media/base/video_adapter.cc:343
↓ 36 callersMethodParseString
logging/rtc_event_log/rtc_event_log_parser.cc:1096
↓ 36 callersMethodSetConfiguration
pc/peer_connection.cc:3891
↓ 36 callersMethodSetRates
modules/video_coding/codecs/vp9/vp9_impl.cc:416
↓ 36 callersMethodSetType
p2p/base/mdns_message.cc:139
↓ 36 callersFunctionTempFilename
Generate a temporary filename in a safe way. Largely copied from talk/base/{unixfilesystem,win32filesystem}.cc.
test/testsupport/file_utils.cc:104
↓ 36 callersMethodempty
api/array_view.h:101
↓ 36 callersMethodforward
test/scenario/call_client.h:165
↓ 36 callersMethodmax
modules/audio_processing/echo_detector/moving_max.cc:43
↓ 36 callersMethodpayload_size
modules/audio_coding/test/TestStereo.cc:79
↓ 36 callersMethodport
p2p/base/turn_port.cc:159
↓ 36 callersFunctionstr
Returns a pointer to the built string. The name |str()| is borrowed for compatibility reasons as we replace usage of stringstream throughout the code
rtc_base/strings/string_builder.h:51
↓ 35 callersMethodAdaptFrame
media/base/video_adapter_unittest.cc:80
↓ 35 callersMethodAddRule
rtc_base/firewall_socket_server.cc:144
↓ 35 callersFunctionCompactNtp
Helper function for compact ntp representation: RFC 3550, Section 4. Time Format. Wallclock time is represented using the timestamp format of the Netw
modules/rtp_rtcp/source/time_util.h:42
↓ 35 callersFunctionCreateBuiltinAudioEncoderFactory
api/audio_codecs/builtin_audio_encoder_factory.cc:57
↓ 35 callersMethodEncodeBatch
logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc:247
↓ 35 callersFunctionExtractNativePC
sdk/android/src/jni/pc/peer_connection.cc:65
↓ 35 callersFunctionGetFirstAudioContent
pc/media_session.cc:2763
↓ 35 callersMethodGetStats
media/engine/fake_webrtc_call.cc:61
↓ 35 callersMethodInitEncode
TODO(eladalon): s/inst/codec_settings/g.
modules/video_coding/codecs/vp9/vp9_impl.cc:445
↓ 35 callersMethodMoveValue
Moves our current value out of this object and returns it, or DCHECK-fails if !this->ok().
api/rtc_error.h:319
↓ 35 callersMethodMutableDataU
api/video/i010_buffer.cc:192
↓ 35 callersMethodMutableDataV
api/video/i010_buffer.cc:195
↓ 35 callersMethodRand<bool>
rtc_base/random.cc:64
↓ 35 callersMethodReadBits
modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc:55
↓ 35 callersMethodReceiver
call/call.cc:625
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