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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 28 callersMethodSetMediaChannel
pc/rtp_sender.cc:129
↓ 28 callersFunctionTestSetPacketLossRate
Sets the packet loss rate to each number in the vector in turn, and verifies that the loss rate as reported by the encoder is |expected_return| for al
modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc:243
↓ 28 callersMethodice_connection_state
pc/peer_connection_integrationtest.cc:532
↓ 28 callersMethodid
()
sdk/android/api/org/webrtc/RtpSender.java:86
↓ 28 callersMethodmin
modules/audio_processing/aec3/api_call_jitter_metrics.h:25
↓ 28 callersMethodrtcp_mux_enabled
pc/composite_rtp_transport.cc:108
↓ 28 callersMethodsample_rate
modules/audio_device/android/aaudio_wrapper.cc:319
↓ 28 callersMethodsamples_per_channel
api/audio/audio_frame.h:102
↓ 28 callersMethodsender_report
test/rtcp_packet_parser.h:101
↓ 28 callersMethodsize
api/video/video_frame.cc:297
↓ 28 callersFunctiontokenize
rtc_base/string_encode.cc:155
↓ 28 callersMethodwants
media/base/video_broadcaster.cc:53
↓ 27 callersMethodAvg
video/send_statistics_proxy.cc:1333
↓ 27 callersMethodClear
modules/audio_coding/neteq/audio_vector.cc:36
↓ 27 callersFunctionConnect
TODO(deadbeef): Deprecated. Remove when Chromium's IceTransportChannel does not depend on this.
p2p/base/p2p_transport_channel.h:115
↓ 27 callersFunctionCreateIceConfig
p2p/base/p2p_transport_channel_unittest.cc:128
↓ 27 callersMethodEncodedImage
api/video/encoded_image.cc:64
↓ 27 callersMethodGetI420
api/video/video_frame_buffer.cc:17
↓ 27 callersMethodGetSize
rtc_base/memory_stream.cc:100
↓ 27 callersMethodIsRecoveryComplete
modules/rtp_rtcp/source/rtp_fec_unittest.cc:149
↓ 27 callersMethodIsSrtpActive
pc/srtp_transport.cc:351
↓ 27 callersMethodOnRatesUpdated
modules/video_coding/codecs/vp8/screenshare_layers.cc:248
↓ 27 callersMethodProcessReverseStream
modules/audio_processing/audio_processing_impl.cc:1414
↓ 27 callersMethodSend
rtc_base/thread.cc:822
↓ 27 callersMethodSetIceConfig
p2p/base/basic_ice_controller.cc:68
↓ 27 callersFunctionSetSettingIfSpecified
modules/audio_processing/test/audioproc_float_impl.cc:280
↓ 27 callersMethodUpdateStats
pc/stats_collector.cc:614
↓ 27 callersMethodcend
rtc_base/buffer.h:199
↓ 27 callersMethoddata
sdk/android/src/jni/encoded_image.cc:36
↓ 27 callersMethodfirst_ssrc
media/base/stream_params.h:109
↓ 27 callersMethodgetDisplaySize
Calculate display size based on scaling type, video aspect ratio, and maximum display size.
sdk/android/api/org/webrtc/RendererCommon.java:205
↓ 27 callersMethodgetName
()
sdk/android/api/org/webrtc/VideoCodecInfo.java:72
↓ 27 callersMethodpadding_size
modules/rtp_rtcp/source/rtp_packet.h:66
↓ 27 callersMethodrotation
media/base/fake_video_renderer.h:44
↓ 27 callersMethodseconds
api/units/timestamp.h:47
↓ 27 callersMethodsend_codecs
media/base/fake_media_engine.cc:603
↓ 27 callersMethodset_codecs
pc/session_description.h:301
↓ 27 callersFunctionswap
Swaps two buffers.
rtc_base/copy_on_write_buffer.h:249
↓ 27 callersMethodupdated_region
modules/desktop_capture/desktop_frame_generator.cc:171
↓ 26 callersMethodAddSample
modules/congestion_controller/bbr/data_transfer_tracker_unittest.cc:28
↓ 26 callersMethodAddTrack
pc/peer_connection_integrationtest.cc:360
↓ 26 callersMethodApply
rtc_base/numerics/exp_filter.cc:24
↓ 26 callersMethodCopyFrom
modules/audio_processing/audio_buffer.cc:114
↓ 26 callersMethodCreateRoutes
test/scenario/scenario.cc:128
↓ 26 callersFunctionDesktopFrameDataEquals
modules/desktop_capture/test_utils.cc:30
↓ 26 callersFunctionDetectOptimization
modules/audio_processing/aec3/aec3_common.cc:21
↓ 26 callersMethodEndOfFile
modules/audio_coding/test/RTPFile.cc:127
↓ 26 callersMethodGetNackList
modules/audio_coding/neteq/neteq_impl.cc:458
↓ 26 callersMethodGetSecondBest
modules/congestion_controller/bbr/windowed_filter.h:149
↓ 26 callersMethodListen
rtc_base/async_socket.cc:78
↓ 26 callersMethodLookup
modules/pacing/paced_sender_unittest.cc:54
↓ 26 callersMethodMaybeLogEncoderConfig
modules/audio_coding/audio_network_adaptor/event_log_writer.cc:41
↓ 26 callersFunctionPayloadStringToCodecType
api/video_codecs/video_codec.cc:165
↓ 26 callersMethodRemainingBitCount
rtc_base/bit_buffer.cc:82
↓ 26 callersMethodSendCompoundPacket
modules/rtp_rtcp/source/rtcp_transceiver.cc:88
↓ 26 callersMethodSetCandidateFilter
p2p/base/port_allocator.cc:287
↓ 26 callersMethodStart
pc/sctp_transport.cc:102
↓ 26 callersMethodToCodecParameters
media/base/codec.cc:191
↓ 26 callersMethodTranslate
modules/desktop_capture/desktop_region.cc:370
↓ 26 callersMethodas_rtp_data
pc/session_description.h:80
↓ 26 callersMethodclear_negotiation_needed
pc/test/mock_peer_connection_observers.h:218
↓ 26 callersMethodcreateInitializationOptions
()
sdk/android/api/org/webrtc/PeerConnectionFactory.java:125
↓ 26 callersMethodextmap_allow_mixed_enum
pc/session_description.h:229
↓ 26 callersMethodframes_per_buffer
modules/audio_device/include/audio_device_defines.h:107
↓ 26 callersMethodinitialize
Loads and initializes WebRTC. This must be called at least once before creating a PeerConnectionFactory. Replaces all the old initialization methods.
sdk/android/api/org/webrtc/PeerConnectionFactory.java:303
↓ 26 callersMethodlatest_bitrate
modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h:44
↓ 26 callersMethodsize
api/test/network_emulation/network_emulation_interfaces.h:38
↓ 26 callersMethodstop
Stops the capture. Waits until no more calls to capture observer will be made. If waitCameraStop is true, also waits for the camera to stop.
sdk/android/src/java/org/webrtc/CameraSession.java:40
↓ 25 callersFunctionABSL_FLAG
modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc:23
↓ 25 callersFunctionAdd
NOLINT
rtc_base/numerics/mod_ops.h:22
↓ 25 callersMethodAddInt
api/stats_types.cc:756
↓ 25 callersMethodAddReceivedPacket
modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc:310
↓ 25 callersMethodAddSamples
rtc_base/rate_tracker.cc:110
↓ 25 callersMethodAddTransportInfo
pc/session_description.cc:196
↓ 25 callersMethodAddVideoTrack
pc/peer_connection_integrationtest.cc:324
↓ 25 callersMethodAddVoiceChannel
pc/test/fake_peer_connection_for_stats.h:135
↓ 25 callersMethodAdvanceTime
test/time_controller/real_time_controller.cc:68
↓ 25 callersMethodAllocatePayload
modules/rtp_rtcp/source/rtp_packet.cc:390
↓ 25 callersMethodAppendExpectedBitstream
video/rtp_video_stream_receiver_unittest.cc:115
↓ 25 callersMethodBuild
api/test/network_emulation_manager.cc:58
↓ 25 callersFunctionComputeHmac
Compute a RFC 2104 HMAC: H(K XOR opad, H(K XOR ipad, text))
rtc_base/message_digest.cc:97
↓ 25 callersFunctionCreatePacket
modules/video_coding/loss_notification_controller_unittest.cc:36
↓ 25 callersMethodDataY
common_video/video_frame_buffer.cc:50
↓ 25 callersMethodGetStartBitrate
call/bitrate_allocator.cc:542
↓ 25 callersMethodGetTaskQueueFactory
test/time_controller/real_time_controller.cc:44
↓ 25 callersMethodH264
api/video_codecs/video_codec.cc:122
↓ 25 callersFunctionJavaToStdString
Deprecated. Use JavaToNativeString.
sdk/android/native_api/jni/java_types.h:330
↓ 25 callersMethodMakeAudioDecoder
api/audio_codecs/test/audio_decoder_factory_template_unittest.cc:66
↓ 25 callersMethodProcessAndGetStats
video/stats_counter.cc:194
↓ 25 callersFunctionRunTest
test/run_test.cc:16
↓ 25 callersMethodSet
modules/audio_processing/include/config.h:126
↓ 25 callersMethodSetReceiveCodecs
modules/audio_coding/acm2/audio_coding_module.cc:576
↓ 25 callersMethodSignalChannelNetworkState
call/call.cc:1108
↓ 25 callersFunctionToString
rtc_base/string_encode.cc:294
↓ 25 callersMethodUpdate
Updates best estimates with |sample|, and expires and updates best estimates as necessary.
modules/congestion_controller/bbr/windowed_filter.h:92
↓ 25 callersMethodUpdateConfiguration
modules/video_coding/codecs/vp8/screenshare_layers.cc:497
↓ 25 callersMethodchannel_send
audio/audio_send_stream_unittest.cc:199
↓ 25 callersMethodgetApplicationContext
Returns the stored application context. @deprecated crbug.com/webrtc/8937
rtc_base/java/src/org/webrtc/ContextUtils.java:41
↓ 25 callersMethodipv6_address
rtc_base/ip_address.cc:116
↓ 25 callersMethodsample_rate
modules/audio_processing/test/conversational_speech/multiend_call.h:66
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