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Functions33,939 in github.com/HackWebRTC/webrtc

⨍ Functions 33,939◇ Types & classes 7,371↳ Endpoints 3
↓ 25 callersMethodsending
media/base/fake_media_engine.h:61
↓ 25 callersMethodtarget_rate
modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc:299
↓ 25 callersMethodwidth
video/video_stream_encoder_unittest.cc:102
↓ 24 callersMethodAddLostPacket
modules/rtp_rtcp/source/packet_loss_stats.cc:32
↓ 24 callersMethodAddSample
video/overuse_frame_detector.cc:188
↓ 24 callersFunctionAddSink
TODO(tommi): Make pure virtual.
api/media_stream_interface.h:254
↓ 24 callersMethodAddSource
modules/audio_mixer/audio_mixer_impl.cc:160
↓ 24 callersFunctionConfigure
modules/video_coding/codecs/vp9/svc_rate_allocator_unittest.cc:24
↓ 24 callersFunctionCreatePacket
modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc:66
↓ 24 callersMethodDataU
common_video/video_frame_buffer.cc:52
↓ 24 callersMethodDataV
common_video/video_frame_buffer.cc:54
↓ 24 callersMethodInsertPacket
modules/audio_coding/neteq/neteq_impl_unittest.cc:1622
↓ 24 callersMethodIsSupportedDecoder
api/audio_codecs/audio_decoder_factory_template.h:32
↓ 24 callersFunctionJavaToNativeOptionalInt
sdk/android/native_api/jni/java_types.cc:143
↓ 24 callersMethodMakeAudioEncoder
api/audio_codecs/test/audio_encoder_factory_template_unittest.cc:66
↓ 24 callersMethodProcess
call/degraded_call.cc:56
↓ 24 callersMethodRand
modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc:52
↓ 24 callersMethodSetConfiguration
Deprecated
p2p/base/port_allocator.cc:120
↓ 24 callersMethodSetDirection
pc/rtp_transceiver.cc:281
↓ 24 callersMethodSetReceiver
video/video_analyzer.cc:161
↓ 24 callersMethodToMs
system_wrappers/include/ntp_time.h:38
↓ 24 callersMethodTotalBytes
modules/rtp_rtcp/include/rtp_rtcp_defines.h:312
↓ 24 callersMethodUpdate
modules/audio_coding/neteq/delay_manager.cc:130
↓ 24 callersFunctionValidatePayloadType
Flag validators.
modules/audio_coding/neteq/tools/neteq_rtpplay.cc:191
↓ 24 callersFunctionVerifyFramesAreEqual
audio/utility/audio_frame_operations_unittest.cc:60
↓ 24 callersMethodWriteBits
modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.cc:199
↓ 24 callersMethodcomponent
p2p/base/port.h:225
↓ 24 callersMethodconnected
p2p/base/turn_port.h:142
↓ 24 callersFunctiondata
Data buffer used for the frame.
modules/desktop_capture/desktop_frame.h:59
↓ 24 callersMethoddtls_transport
pc/sctp_transport.cc:58
↓ 24 callersMethodfiltered
Returns current filtered value.
rtc_base/numerics/exp_filter.h:36
↓ 24 callersMethodgetEglBaseContext
()
sdk/android/api/org/webrtc/EglBase.java:237
↓ 24 callersMethodheight
video/video_stream_encoder_unittest.cc:103
↓ 24 callersMethodlocal_streams
pc/peer_connection.cc:1429
↓ 24 callersMethodnum_channels
modules/audio_processing/agc/agc_manager_direct.h:64
↓ 24 callersMethodnum_channels
common_audio/wav_file.h:64
↓ 24 callersMethodport
p2p/base/port_unittest.cc:363
↓ 24 callersMethodrend
logging/rtc_event_log/rtc_event_log_parser.h:185
↓ 24 callersMethodsetEnabled
(boolean enable)
sdk/android/api/org/webrtc/MediaStreamTrack.java:97
↓ 24 callersMethodset_direction
Sets the intended direction for this transceiver. Intended to be used internally over SetDirection since this does not trigger a negotiation needed ca
pc/rtp_transceiver.h:138
↓ 24 callersMethodstopped
pc/rtp_transceiver.cc:273
↓ 23 callersMethodAdd
rtc_base/numerics/sample_counter.cc:23
↓ 23 callersMethodAddLayerWithAlternatives
pc/simulcast_description.cc:32
↓ 23 callersMethodAddMessageIntegrity
api/transport/stun.cc:292
↓ 23 callersMethodAddVideoChannel
pc/test/fake_peer_connection_for_stats.h:153
↓ 23 callersFunctionCheckException
sdk/android/src/jni/jni_generator_helper.h:93
↓ 23 callersMethodComputeRate
Computes the average rate over the rate tracker's recording interval of bucket_milliseconds * bucket_count.
rtc_base/rate_tracker.h:34
↓ 23 callersMethodCreateAudioSendStream
audio/audio_send_stream_unittest.cc:182
↓ 23 callersFunctionCreateRandomId
rtc_base/helpers.cc:194
↓ 23 callersMethodData
api/call/audio_sink.h:30
↓ 23 callersMethodFlushAsyncInvokes
api/test/loopback_media_transport.cc:205
↓ 23 callersMethodGet
modules/audio_device/android/opensles_common.h:45
↓ 23 callersMethodGet
sdk/android/src/jni/audio_device/opensles_common.h:50
↓ 23 callersMethodGetAllLayers
pc/simulcast_description.cc:48
↓ 23 callersMethodGetGroupByName
pc/session_description.cc:253
↓ 23 callersMethodGetSctpTransport
pc/peer_connection.cc:4363
↓ 23 callersMethodGetStatistician
modules/rtp_rtcp/source/receive_statistics_impl.cc:321
↓ 23 callersMethodGetVariance
Returns unbiased sample variance in O(1) time.
rtc_base/numerics/running_statistics.h:134
↓ 23 callersFunctionI420PSNR
Compute PSNR for an I420 frame (all planes). Can upscale test frame.
common_video/libyuv/webrtc_libyuv.cc:283
↓ 23 callersMethodInjectData
test/pc/e2e/analyzer/video/default_encoded_image_data_injector.cc:39
↓ 23 callersMethodInsertEvent
Inserts a DTMF event into the buffer. The event should be parsed from the bit stream using the ParseEvent method above before inserting it in the buff
modules/audio_coding/neteq/dtmf_buffer.cc:99
↓ 23 callersMethodInsertPacket
modules/audio_coding/neteq/neteq_unittest.cc:614
↓ 23 callersFunctionIsMediaContentOfType
pc/media_session.cc:100
↓ 23 callersMethodNetworkReceivedPackets
modules/rtp_rtcp/source/rtp_fec_unittest.cc:99
↓ 23 callersMethodNextFrame
test/fake_encoder.cc:166
↓ 23 callersMethodNextPacket
modules/audio_coding/neteq/packet_buffer_unittest.cc:76
↓ 23 callersMethodProcess
modules/remote_bitrate_estimator/remote_estimator_proxy.cc:149
↓ 23 callersMethodProcess
modules/video_coding/nack_module.cc:210
↓ 23 callersFunctionProfileLevelIdToString
media/base/h264_profile_level_id.cc:213
↓ 23 callersMethodPushBack
modules/audio_coding/neteq/sync_buffer.cc:23
↓ 23 callersMethodRead
rtc_base/memory/fifo_buffer.cc:87
↓ 23 callersFunctionRemoveSink
api/media_stream_interface.h:187
↓ 23 callersMethodSetCodecSettings
modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc:152
↓ 23 callersFunctionSetFrom
api/audio_options.cc:29
↓ 23 callersMethodSetSink
media/engine/fake_webrtc_call.cc:107
↓ 23 callersMethodState
modules/remote_bitrate_estimator/overuse_detector.cc:80
↓ 23 callersMethodToSensitiveString
rtc_base/ip_address.cc:139
↓ 23 callersFunctionTransformDbMetricForReporting
modules/audio_processing/aec3/echo_remover_metrics.cc:231
↓ 23 callersMethodWaitForEncodedFrame
video/video_stream_encoder_unittest.cc:655
↓ 23 callersFunctionWebRtcIsac_EncHistMulti
* code symbols into arithmetic bytestream */
modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c:18
↓ 23 callersMethodbandwidth
pc/session_description.h:124
↓ 23 callersMethodcount
modules/rtp_rtcp/source/rtcp_packet/common_header.h:32
↓ 23 callersMethodend
sdk/android/native_api/jni/java_types.h:102
↓ 23 callersMethodfmt
Depending on packet type same header field can be used either as count or as feedback message type (fmt). Caller expected to know how it is used.
modules/rtp_rtcp/source/rtcp_packet/common_header.h:31
↓ 23 callersFunctionheight
modules/desktop_capture/desktop_geometry.h:109
↓ 23 callersMethodmakeCurrent
()
sdk/android/api/org/webrtc/EglBase.java:249
↓ 23 callersMethodnum_pings_sent
p2p/base/connection.h:271
↓ 23 callersMethodport
pc/session_description.h:393
↓ 23 callersMethodreceived_packets
media/engine/fake_webrtc_call.h:91
↓ 23 callersMethodsetNativeLibraryName
(String nativeLibraryName)
sdk/android/api/org/webrtc/PeerConnectionFactory.java:114
↓ 23 callersMethodset_capture_time_ms
modules/rtp_rtcp/source/rtp_packet_to_send.h:49
↓ 23 callersMethodset_stream_delay_ms
modules/audio_processing/audio_processing_impl.cc:1552
↓ 23 callersFunctionwidth
modules/desktop_capture/desktop_geometry.h:108
↓ 22 callersMethodAddSendStream
media/engine/webrtc_voice_engine.cc:1763
↓ 22 callersMethodAddSender
pc/rtp_transceiver.cc:162
↓ 22 callersMethodBuffer
Buffer.
modules/rtp_rtcp/source/rtp_packet.h:75
↓ 22 callersMethodCheckForOveruse
video/overuse_frame_detector.cc:634
↓ 22 callersFunctionCompactNtpRttToMs
modules/rtp_rtcp/source/time_util.cc:77
↓ 22 callersMethodCreateAudioTrack
pc/peer_connection_wrapper.cc:275
↓ 22 callersFunctionCreateBuiltinVideoEncoderFactory
api/video_codecs/builtin_video_encoder_factory.cc:81
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