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Functions33,939 in github.com/HackWebRTC/webrtc

↓ 19 callersMethodcumulative_lost_signed
modules/rtp_rtcp/source/rtcp_packet/report_block.h:52
↓ 19 callersMethodenabled
pc/channel.h:108
↓ 19 callersMethodfull_sync
modules/video_coding/decoding_state.cc:169
↓ 19 callersMethodgetDataY
Returns a direct ByteBuffer containing Y-plane data. The buffer capacity is at least getStrideY() getHeight() bytes. The position of the returned buff
sdk/android/api/org/webrtc/VideoFrame.java:72
↓ 19 callersMethodid
A unique ID assigned when the connection is created.
p2p/base/connection.h:89
↓ 19 callersMethodpacket_overhead
modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h:38
↓ 19 callersMethodremote_address
rtc_base/test_client.h:57
↓ 19 callersMethodsender_ssrc
test/rtcp_packet_parser.h:107
↓ 19 callersMethodset
modules/desktop_capture/desktop_geometry.h:66
↓ 19 callersMethodset_audio_codecs
pc/media_session.cc:1354
↓ 19 callersMethodset_send_blocked
Set true to emulate the SCTP stream being blocked by congestion control.
pc/test/fake_data_channel_provider.h:91
↓ 19 callersMethodstats
test/network/network_emulation.cc:320
↓ 19 callersMethodtype
modules/audio_processing/include/audio_processing.h:438
↓ 18 callersMethodAdd
static
rtc_base/thread.cc:121
↓ 18 callersFunctionAddIdIfDefined
pc/rtc_stats_traversal.cc:45
↓ 18 callersMethodAddSample
modules/congestion_controller/bbr/data_transfer_tracker.cc:22
↓ 18 callersFunctionAreEqual
rtc_base/ip_address_unittest.cc:97
↓ 18 callersFunctionCodecSettings
test/video_codec_settings.h:27
↓ 18 callersMethodCodecSpecific
* Get codec specific info. * The returned pointer is only valid as long as the VCMEncodedFrame * is valid. Also, VCMEncodedFrame owns the
modules/video_coding/encoded_frame.h:117
↓ 18 callersFunctionComputeDigest
rtc_base/message_digest.cc:53
↓ 18 callersFunctionConvertToProtoFormat
logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc:78
↓ 18 callersFunctionCopy
Deprecated.
api/video/i420_buffer.h:39
↓ 18 callersMethodCopyAndFixBitstream
modules/video_coding/h264_sps_pps_tracker.cc:47
↓ 18 callersFunctionCreateAimdRateControlStates
modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc:40
↓ 18 callersMethodCreateAnswer
p2p/base/transport_description_factory.cc:63
↓ 18 callersFunctionCreateDevice
modules/audio_device/win/core_audio_utility_win.cc:748
↓ 18 callersMethodCreateTaskQueue
rtc_base/thread_unittest.cc:1143
↓ 18 callersFunctionCreateVideoCodecTestFixture
api/test/create_videocodec_test_fixture.cc:24
↓ 18 callersMethodData
Returns start of unprocessed data.
rtc_base/byte_buffer.h:133
↓ 18 callersMethodEnsureCapacity
Ensure that the buffer size can be increased to at least capacity without further reallocation. (Of course, this operation might need to reallocate th
rtc_base/buffer.h:334
↓ 18 callersMethodEquals
modules/desktop_capture/desktop_region.cc:60
↓ 18 callersFunctionFilterRtpExtensions
media/engine/webrtc_media_engine.cc:87
↓ 18 callersMethodGetApplicationLimitedRegionStartTime
modules/congestion_controller/goog_cc/alr_detector.cc:107
↓ 18 callersMethodGetConfigs
rtc_base/experiments/cpu_speed_experiment.cc:27
↓ 18 callersMethodGetEvent
modules/audio_coding/neteq/dtmf_buffer.cc:119
↓ 18 callersFunctionGetFileSize
test/testsupport/file_utils.cc:234
↓ 18 callersMethodGetMin
Returns minimum among all seen samples, in O(1) time. This isn't affected by RemoveSample().
rtc_base/numerics/running_statistics.h:109
↓ 18 callersMethodGetStunMessage
p2p/base/port.cc:442
↓ 18 callersMethodGetType
p2p/base/mdns_message.cc:152
↓ 18 callersMethodGetVideoChannel
media/base/fake_media_engine.cc:649
↓ 18 callersMethodIsEquivalent
api/candidate.cc:57
↓ 18 callersFunctionIsNull
sdk/android/native_api/jni/java_types.cc:100
↓ 18 callersFunctionIsScreenshare
api/video/video_content_type.cc:80
↓ 18 callersFunctionMUL_ACCUM_1
common_audio/signal_processing/resample_by_2.c:30
↓ 18 callersFunctionMUL_ACCUM_2
common_audio/signal_processing/resample_by_2.c:46
↓ 18 callersFunctionMakeUsageFingerprint
pc/peer_connection_histogram_unittest.cc:64
↓ 18 callersMethodPing
p2p/base/connection.cc:835
↓ 18 callersMethodRemoveGroupByName
pc/session_description.cc:233
↓ 18 callersMethodSend
p2p/base/tcp_port.cc:376
↓ 18 callersMethodSetCodecPreferences
pc/rtp_transceiver.cc:313
↓ 18 callersMethodSetDelayOnAddress
rtc_base/virtual_socket_server.h:115
↓ 18 callersMethodSetDtlsTransports
pc/dtls_srtp_transport.cc:32
↓ 18 callersMethodSetIceRole
p2p/base/p2p_transport_channel_unittest.cc:500
↓ 18 callersMethodSetOption
p2p/base/tcp_port.cc:255
↓ 18 callersMethodSetRtt
modules/rtp_rtcp/source/rtp_packet_history.cc:109
↓ 18 callersMethodSplitIntoFrequencyBands
modules/audio_processing/audio_buffer.cc:364
↓ 18 callersMethodStart
test/call_test.cc:602
↓ 18 callersMethodStop
p2p/base/port_unittest.cc:310
↓ 18 callersFunctionTestEncodingAndDecoding
Encodes |values| based on |base|, then decodes the result and makes sure that it is equal to the original input. If |encoded_string| is non-null, the
logging/rtc_event_log/encoder/delta_encoding_unittest.cc:71
↓ 18 callersMethodTestMemberIsIDReference
pc/rtc_stats_integrationtest.cc:267
↓ 18 callersFunctionTimeDiff
rtc_base/time_utils.cc:231
↓ 18 callersFunctionToSigned
logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h:64
↓ 18 callersFunctionWebRtcOpus_SetBitRate
modules/audio_coding/codecs/opus/opus_interface.cc:200
↓ 18 callersFunctionWebRtcSpl_AddSatW32
common_audio/signal_processing/include/spl_inl.h:85
↓ 18 callersFunctionWebRtcSpl_FilterARFastQ12
common_audio/signal_processing/filter_ar_fast_q12.c:18
↓ 18 callersFunctionWriteStunMessage
p2p/base/port_unittest.cc:125
↓ 18 callersMethodcall
()
rtc_base/java/src/org/webrtc/ThreadUtils.java:197
↓ 18 callersMethodcheckNoGLES2Error
(String msg)
sdk/android/api/org/webrtc/GlUtil.java:32
↓ 18 callersMethodcreateIceServer
()
sdk/android/api/org/webrtc/PeerConnection.java:316
↓ 18 callersMethodend
stats/rtc_stats_report.cc:115
↓ 18 callersMethodipv4_address
rtc_base/ip_address.cc:120
↓ 18 callersMethodlog
(Severity severity, String tag, String message)
rtc_base/java/src/org/webrtc/Logging.java:120
↓ 18 callersMethodmin_video_frames_received_per_track
Takes minimum of video frames received for each track. Can be used like: EXPECT_GE(expected_frames, min_video_frames_received_per_track()); To ensur
pc/peer_connection_integrationtest.cc:422
↓ 18 callersMethodmsid_signaling
pc/session_description.h:608
↓ 18 callersMethodnetwork_cost
p2p/base/port.h:347
↓ 18 callersMethodnetwork_route
pc/rtp_transport_unittest.cc:56
↓ 18 callersMethodnetwork_thread
pc/channel.h:104
↓ 18 callersMethodntp
modules/rtp_rtcp/source/rtcp_packet/rrtr.h:43
↓ 18 callersMethodpacket_number
An identifier of the packet.
logging/rtc_event_log/events/rtc_event_generic_packet_sent.h:35
↓ 18 callersMethodrtp_dtls_transport
pc/jsep_transport.h:185
↓ 18 callersMethodsdes
test/rtcp_packet_parser.h:100
↓ 18 callersMethodsize
rtc_base/virtual_socket_server.cc:76
↓ 18 callersMethodssrc
modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h:36
↓ 18 callersMethodstartListening
Start to stream textures to the given |listener|. If you need to change listener, you need to call stopListening() first.
sdk/android/api/org/webrtc/SurfaceTextureHelper.java:225
↓ 18 callersMethodstream
rtc_base/stream.h:172
↓ 18 callersMethodvalid
modules/audio_processing/test/conversational_speech/multiend_call.h:65
↓ 17 callersMethodAddCName
modules/rtp_rtcp/source/rtcp_packet/sdes.cc:148
↓ 17 callersMethodAddIceCandidate
Adds a new ICE candidate to the first transport.
pc/peer_connection_ice_unittest.cc:67
↓ 17 callersMethodAdjustRateAllocation
video/encoder_bitrate_adjuster.cc:73
↓ 17 callersMethodAdvance
modules/desktop_capture/desktop_region.cc:518
↓ 17 callersMethodAllocate
modules/video_coding/utility/simulcast_rate_allocator.cc:68
↓ 17 callersFunctionAllocationLimitsEq
call/bitrate_allocator_unittest.cc:29
↓ 17 callersMethodComputeDeltas
modules/remote_bitrate_estimator/inter_arrival.cc:33
↓ 17 callersMethodConfigureEncoder
video/video_stream_encoder_unittest.cc:433
↓ 17 callersFunctionCreateBuiltinVideoBitrateAllocatorFactory
api/video/builtin_video_bitrate_allocator_factory.cc:54
↓ 17 callersFunctionCreateClient
modules/audio_device/win/core_audio_utility_win.cc:874
↓ 17 callersFunctionCreateFakeCandidate
pc/rtc_stats_collector_unittest.cc:175
↓ 17 callersFunctionCreatePeerConnectionFactory
api/create_peerconnection_factory.cc:29
↓ 17 callersMethodCurrentNtpInMilliseconds
test/drifting_clock.cc:45
↓ 17 callersFunctionDivideRoundUp
rtc_base/numerics/divide_round.h:22
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