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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 22 callersFunctionCreateFecControllerPlrBased
modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc:56
↓ 22 callersMethodCreateOffer
p2p/base/transport_description_factory.cc:29
↓ 22 callersMethodCreateRollback
pc/peer_connection_wrapper.cc:128
↓ 22 callersMethodDataCountersUpdated
video/send_statistics_proxy.cc:1236
↓ 22 callersMethodDelay
modules/audio_processing/aec3/render_delay_buffer.cc:56
↓ 22 callersMethodDumpWav
modules/audio_processing/logging/apm_data_dumper.h:238
↓ 22 callersFunctionFloatS16ToS16
common_audio/audio_util.cc:30
↓ 22 callersMethodGetMean
Returns mean in O(1) time.
rtc_base/numerics/running_statistics.h:126
↓ 22 callersMethodGetQpThresholds
rtc_base/experiments/quality_scaling_experiment.cc:61
↓ 22 callersFunctionGetSimulcastConfig
media/engine/simulcast.cc:224
↓ 22 callersMethodGetState
p2p/base/dtls_transport.cc:110
↓ 22 callersMethodHasSimulcast
Simulcast functionality.
pc/session_description.h:237
↓ 22 callersMethodInitEncode
TODO(eladalon): s/inst/codec_settings/g.
modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc:460
↓ 22 callersMethodMinFps
rtc_base/experiments/balanced_degradation_settings.cc:405
↓ 22 callersMethodMix
modules/audio_mixer/audio_mixer_impl.cc:121
↓ 22 callersMethodNumPacketsInBuffer
modules/audio_coding/neteq/packet_buffer.cc:267
↓ 22 callersMethodProcess
modules/audio_processing/test/debug_dump_test.cc:206
↓ 22 callersMethodPushBackInterleaved
modules/audio_coding/neteq/sync_buffer.cc:40
↓ 22 callersMethodQueryAudioEncoder
api/audio_codecs/test/audio_encoder_factory_template_unittest.cc:62
↓ 22 callersMethodReadSamples
common_audio/wav_file.cc:90
↓ 22 callersMethodReadString
rtc_base/byte_buffer.cc:130
↓ 22 callersMethodRemoveRecvStream
media/engine/webrtc_voice_engine.cc:1874
↓ 22 callersFunctionRtpTransceiverDirectionHasSend
pc/rtp_media_utils.cc:30
↓ 22 callersMethodRunTest
Processes all frames in the clip and verifies the result.
modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc:396
↓ 22 callersMethodSetStats
media/engine/fake_webrtc_call.cc:36
↓ 22 callersMethodSizePartitions
Returns the filter size.
modules/audio_processing/aec3/adaptive_fir_filter.h:117
↓ 22 callersMethodStop
modules/audio_device/android/aaudio_wrapper.cc:192
↓ 22 callersMethodallocator
p2p/client/basic_port_allocator.cc:299
↓ 22 callersMethodcomplete_count
p2p/base/port_unittest.cc:274
↓ 22 callersMethodlatest_bitrate
modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h:40
↓ 22 callersMethodnum_bands
modules/audio_processing/audio_buffer.h:69
↓ 22 callersMethodparams
test/pc/e2e/test_peer.h:76
↓ 22 callersMethodreset
()
examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java:197
↓ 22 callersMethodset_enabled
pc/video_track.cc:71
↓ 22 callersMethodset_stream_analog_level
TODO(ajm): ensure this is called under kAdaptiveAnalog.
modules/audio_processing/gain_control_impl.cc:282
↓ 22 callersMethodtime_controller
test/network/network_emulation_manager.h:86
↓ 22 callersMethodtype
modules/rtp_rtcp/source/rtcp_packet/common_header.h:28
↓ 22 callersMethodupdateSummaryB
(SharedPreferences sharedPreferences, String key)
examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java:270
↓ 22 callersMethodupdated
modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h:42
↓ 21 callersFunctionAbsSendTime
Helper to convert some time format to resolution used in absolute send time header extension, rounded upwards. |t| is the time to convert, in some res
modules/congestion_controller/receive_side_congestion_controller_unittest.cc:34
↓ 21 callersMethodAdd
video/receive_statistics_proxy.cc:927
↓ 21 callersMethodAdd
video/send_statistics_proxy.cc:1328
↓ 21 callersMethodAddDlrrItem
modules/rtp_rtcp/source/rtcp_packet/dlrr.h:56
↓ 21 callersMethodAddTargetBitrate
modules/rtp_rtcp/source/rtcp_packet/target_bitrate.cc:90
↓ 21 callersFunctionAppendFormat
rtc_base/checks.cc:40
↓ 21 callersMethodCanUse
rtc_base/data_rate_limiter.cc:15
↓ 21 callersMethodCheckNextPacket
rtc_base/test_client.cc:97
↓ 21 callersMethodClose
rtc_base/stream.cc:96
↓ 21 callersFunctionCreateBuiltinVideoDecoderFactory
api/video_codecs/builtin_video_decoder_factory.cc:19
↓ 21 callersFunctionCreateController
modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc:49
↓ 21 callersFunctionCreateSimStreamParams
media/base/test_utils.cc:20
↓ 21 callersMethodDropFrame
This function signals when to drop frames to the caller. It makes use of the drop_ratio_ to smooth out the drops over time.
modules/video_coding/utility/frame_dropper.cc:174
↓ 21 callersMethodEncodeFec
modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc:92
↓ 21 callersMethodEqualIPs
rtc_base/socket_address.cc:236
↓ 21 callersFunctionExtractSsrcStatsValue
pc/stats_collector_unittest.cc:257
↓ 21 callersMethodFindValue
api/stats_types.cc:780
↓ 21 callersMethodGetNextBlock
modules/audio_coding/neteq/tools/audio_loop.cc:48
↓ 21 callersMethodGetPercentile
rtc_base/numerics/samples_stats_counter.cc:45
↓ 21 callersMethodGetTransportDescriptionByName
pc/session_description.h:575
↓ 21 callersMethodIsProbing
modules/pacing/bitrate_prober.cc:77
↓ 21 callersMethodIsRegistered
modules/rtp_rtcp/include/rtp_header_extension_map.h:42
↓ 21 callersMethodOnSetEncoderTargetRate
video/send_statistics_proxy.cc:795
↓ 21 callersMethodProcessReceivedFec
TODO(nisse): Drop always-zero return value.
modules/rtp_rtcp/source/ulpfec_receiver_impl.cc:150
↓ 21 callersMethodRelease
media/engine/encoder_simulcast_proxy.cc:30
↓ 21 callersMethodSetBitrates
modules/congestion_controller/goog_cc/probe_controller.cc:144
↓ 21 callersMethodSetEncoder
Utility method for simply replacing the existing encoder with a new one.
modules/audio_coding/include/audio_coding_module.h:108
↓ 21 callersMethodSetExtHighestSeqNum
modules/rtp_rtcp/source/rtcp_packet/report_block.h:41
↓ 21 callersMethodSetIP
rtc_base/socket_address.cc:90
↓ 21 callersMethodSetIceParameters
p2p/base/port.cc:232
↓ 21 callersMethodSetRtpPacketTransport
pc/rtp_transport.cc:50
↓ 21 callersFunctionSetSettingIfFlagSet
modules/audio_processing/test/audioproc_float_impl.cc:301
↓ 21 callersMethodSlice
Sets the |slice| pointers to the |start_frame| position for each channel. Returns |slice| for convenience.
common_audio/channel_buffer.h:105
↓ 21 callersMethodStart
p2p/base/port_unittest.cc:279
↓ 21 callersMethodTriggerCpuNormalUsage
video/video_stream_encoder_unittest.cc:190
↓ 21 callersFunctionUpdateNetworkMetrics
modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc:22
↓ 21 callersMethodWrite
p2p/base/mdns_message.cc:122
↓ 21 callersMethodWrite
rtc_base/memory/fifo_buffer.cc:113
↓ 21 callersFunction_mm_castps_si128
modules/audio_processing/utility/ooura_fft_sse2.cc:30
↓ 21 callersFunction_mm_castsi128_ps
modules/audio_processing/utility/ooura_fft_sse2.cc:27
↓ 21 callersMethodcount
api/jsep_ice_candidate.cc:42
↓ 21 callersMethodcreate
Construct a new SurfaceTextureHelper sharing OpenGL resources with |sharedContext|. A dedicated thread and handler is created for handling the Surface
sdk/android/api/org/webrtc/SurfaceTextureHelper.java:59
↓ 21 callersMethodexpectSignalingChange
(SignalingState newState)
sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java:162
↓ 21 callersMethodhandle
rtc_base/win32_window.h:29
↓ 21 callersMethodhas_ssrcs
pc/session_description.h:197
↓ 21 callersFunctionid
Ownership of |id| is passed to |this|.
api/stats_types.h:369
↓ 21 callersMethodnumber_of_frames
rtc_tools/frame_analyzer/video_temporal_aligner.cc:123
↓ 21 callersMethodoptions
pc/local_audio_source.h:33
↓ 21 callersMethodpc_factory
pc/peer_connection_wrapper.cc:53
↓ 21 callersMethodplayout
media/base/fake_media_engine.h:62
↓ 21 callersMethodrecv_codecs
media/base/fake_media_engine.cc:607
↓ 21 callersMethodrtp_data_channel
pc/peer_connection.h:276
↓ 21 callersMethodsample_rate
sdk/android/src/jni/audio_device/aaudio_wrapper.cc:319
↓ 21 callersMethodsamples_per_channel
modules/audio_processing/include/audio_frame_view.h:44
↓ 21 callersMethodset_component
p2p/base/port.h:226
↓ 21 callersMethodsharedPrefGetBoolean
Get a value from the shared preference or from the intent, if it does not exist the default is used.
examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java:323
↓ 21 callersMethodsignaling_thread
pc/channel.h:197
↓ 21 callersMethodssrcs
Utility accessor for clients that are only interested in ssrc numbers.
media/base/media_channel.h:375
↓ 21 callersMethodupdated
modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h:38
↓ 21 callersMethodwaitForAllExpectationsToBeSatisfied
(int timeoutSeconds)
sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java:510
↓ 20 callersMethodAddFingerprint
pc/webrtc_sdp_unittest.cc:1695
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