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Types & classes7,371 in github.com/HackWebRTC/webrtc

↓ 300 callersClassSocketAddress
rtc_base/http_common.h:19
↓ 111 callersClassBuilder
sdk/android/api/org/webrtc/EncodedImage.java:131
↓ 105 callersClassVideoCodec
modules/video_coding/include/video_codec_initializer.h:23
↓ 76 callersClassSimulcastLayer
Describes a Simulcast Layer. Each simulcast layer has a rid as the identifier and a paused flag. See also: https://tools.ietf.org/html/draft-ietf-mmus
pc/simulcast_description.h:23
↓ 75 callersClassVideoBitrateAllocationParameters
api/video/video_bitrate_allocator.h:19
↓ 73 callersClassDesktopSize
Type used to represent screen/window size.
modules/desktop_capture/desktop_geometry.h:52
↓ 73 callersClassDesktopVector
A vector in the 2D integer space. E.g. can be used to represent screen DPI.
modules/desktop_capture/desktop_geometry.h:21
↓ 59 callersClassDataBuffer
At the JavaScript level, data can be passed in as a string or a blob, so this structure's |binary| flag tells whether the data should be interpreted a
api/data_channel_interface.h:69
↓ 57 callersClassAudioCodec
media/base/codec.h:116
↓ 57 callersClassFeedbackParam
media/base/codec.h:29
↓ 57 callersClassPacedPacketInfo
modules/rtp_rtcp/source/rtp_rtcp_impl.h:42
↓ 51 callersClassBuiltInNetworkBehaviorConfig
BuiltInNetworkBehaviorConfig is a built-in network behavior configuration for built-in network behavior that will be used by WebRTC if no custom Netwo
api/test/simulated_network.h:49
↓ 51 callersClassEchoCanceller3Config
modules/audio_processing/aec3/clockdrift_detector.h:20
↓ 49 callersClassVideoCodecConfig
api/test/peerconnection_quality_test_fixture.h:328
↓ 45 callersClassCryptoOptions
CryptoOptions defines advanced cryptographic settings for native WebRTC. These settings must be passed into RTCConfiguration. WebRTC is secur by defau
sdk/android/api/org/webrtc/CryptoOptions.java:20
↓ 42 callersClassCopyOnWriteBuffer
pc/sctp_utils.h:21
↓ 38 callersClassRTCOfferAnswerOptions
See: https://www.w3.org/TR/webrtc/#idl-def-rtcofferansweroptions
api/peer_connection_interface.h:675
↓ 36 callersClassSdpVideoFormat
api/video_codecs/video_decoder_factory.h:23
↓ 36 callersClassStreamDataCounters
Data usage statistics for a (rtp) stream.
modules/rtp_rtcp/include/rtp_rtcp_defines.h:323
↓ 34 callersClassSdpVideoFormat
modules/video_coding/codecs/h264/include/h264.h:24
↓ 33 callersClassVideoOptions
Options that can be applied to a VideoMediaChannel or a VideoMediaEngine. Used to be flags, but that makes it hard to selectively apply options. We ar
media/base/media_channel.h:105
↓ 32 callersClassNetworkControlUpdate
Contains updates of network controller comand state. Using optionals to indicate whether a member has been updated. The array of probe clusters should
api/transport/network_types.h:226
↓ 30 callersClassCallClientConfig
test/scenario/scenario_config.h:53
↓ 30 callersClassConfig
pc/jsep_transport_controller.h:77
↓ 30 callersEnumFlags
pc/channel_unittest.cc:108
↓ 29 callersClassObserver
pc/jsep_transport_controller.h:52
↓ 28 callersEnumError
pc/srtp_filter.h:42
↓ 28 callersClassFrameLengthChange
modules/audio_coding/audio_network_adaptor/frame_length_controller.h:31
↓ 27 callersClassRTCConfiguration
sdk/objc/api/peerconnection/RTCPeerConnection.h:15
↓ 26 callersClassObu
modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc:110
↓ 25 callersClassRemoteCandidate
Adds the port on which the candidate originated.
p2p/base/p2p_transport_channel.h:70
↓ 24 callersClassMBErr
tools_webrtc/mb/mb.py:1207
↓ 24 callersClassPacketOptions
pc/rtp_transport.h:24
↓ 22 callersClassConfig
modules/audio_coding/audio_network_adaptor/dtx_controller.h:23
↓ 22 callersClassDesktopRegion
modules/desktop_capture/win/dxgi_texture.h:24
↓ 22 callersClassMediaTransportConfig
Media transport config is made available to both transport and audio / video layers, but access to individual interfaces should not be open without ne
api/transport/media/media_transport_config.h:23
↓ 22 callersClassTransportDescription
p2p/base/transport_description.h:121
↓ 22 callersClassTransportOptions
p2p/base/transport_description_factory.h:26
↓ 22 callersEnumWrap
common_audio/ring_buffer.h:25
↓ 21 callersEnumResult
modules/desktop_capture/desktop_capturer.h:35
↓ 20 callersEnumChannels
Note: Do not reorder or reassign these values; other code depends on their ordering to operate correctly. E.g., CoreAudio channel layout computations.
api/audio/channel_layout.h:128
↓ 20 callersClassEmulatedEndpointConfig
api/test/network_emulation_manager.h:45
↓ 20 callersClassTransportInfo
A TransportInfo is NOT a transport-info message. It is comparable to a "ContentInfo". A transport-infos message is basically just a collection of Tra
p2p/base/transport_info.h:27
↓ 18 callersClassNetwork
p2p/base/port_interface.h:24
↓ 18 callersClassProfileLevelId
media/base/h264_profile_level_id.h:49
↓ 18 callersClassProtocolAddress
p2p/base/port.h:116
↓ 18 callersClassStreamConfig
modules/audio_processing/include/audio_processing.h:44
↓ 17 callersClassRidDescription
Description of a Restriction Id (RID) according to: https://tools.ietf.org/html/draft-ietf-mmusic-rid-15 A Restriction Identifier serves two purposes:
media/base/rid_description.h:45
↓ 17 callersClassScalingSettings
Settings for WebRTC quality based scaling.
sdk/android/api/org/webrtc/VideoEncoder.java:121
↓ 16 callersClassConfig
modules/audio_coding/codecs/g711/audio_encoder_pcm.h:23
↓ 16 callersClassNetworkSimulationConfig
TODO(srte): Merge this with BuiltInNetworkBehaviorConfig.
test/scenario/scenario_config.h:218
↓ 16 callersClassSsrcSenderInfo
Information about an SSRC. This data may be locally recorded, or received in an RTCP SR or RR.
media/base/media_channel.h:353
↓ 16 callersClassengine
* Shared state for our app. */
modules/audio_processing/test/android/apmtest/jni/main.c:44
↓ 16 callersClassevent
rtc_base/task_queue_libevent.cc:79
↓ 15 callersInterfaceBuffer
Implements image storage medium. Might be for example an OpenGL texture or a memory region containing I420-data. <p>Reference counting is needed sinc
sdk/android/api/org/webrtc/VideoFrame.java:36
↓ 15 callersClassFakeClock
Fake clock for use with unit tests, which does not tick on its own. Starts at time 0. TODO(deadbeef): Unify with webrtc::SimulatedClock.
rtc_base/fake_clock.h:28
↓ 15 callersClassLevelAndProbability
modules/audio_processing/agc2/vad_with_level.h:22
↓ 15 callersClassNtpTime
system_wrappers/include/ntp_time.h:21
↓ 15 callersClassRgbaColor
A four-byte structure to store a color in BGRA format. This structure also provides functions to be created from uint8_t array, say, DesktopFrame::dat
modules/desktop_capture/rgba_color.h:24
↓ 15 callersClassRtpParameters
media/base/media_channel.h:736
↓ 15 callersClassVectorMath
Provides optimizations for mathematical operations based on vectors.
modules/audio_processing/aec3/vector_math.h:37
↓ 14 callersClassSvcRateAllocator
modules/video_coding/codecs/vp9/svc_rate_allocator.h:26
↓ 13 callersClassSamples
video/stats_counter.h:23
↓ 13 callersClassSendCodecSpec
call/audio_send_stream.h:130
↓ 13 callersClassVideoSimulcastConfig
api/test/peerconnection_quality_test_fixture.h:133
↓ 13 callersClassVideoStreamConfig
test/scenario/scenario_config.h:70
↓ 12 callersClassConfig
test/peer_scenario/peer_scenario_client.h:70
↓ 12 callersClassCryptoParams
pc/channel.h:54
↓ 12 callersClassEncodeDecodeTest
modules/audio_coding/test/EncodeDecodeTest.h:102
↓ 12 callersClassLagEstimate
modules/audio_processing/aec3/matched_filter.h:76
↓ 12 callersClassReceiveTimeInfo
modules/rtp_rtcp/source/rtcp_packet/dlrr.h:22
↓ 11 callersClassFunctorB
rtc_base/thread_unittest.cc:181
↓ 11 callersClassInfo
modules/rtp_rtcp/source/rtp_sequence_number_map.h:36
↓ 11 callersClassLicenseBuilder
tools_webrtc/libs/generate_licenses.py:112
↓ 11 callersClassPccMonitorInterval
PCC divides time into consecutive monitor intervals which are used to test consequences for performance of sending at a certain rate.
modules/congestion_controller/pcc/monitor_interval.h:27
↓ 10 callersClassAudioCodecInfo
Information about how an audio format is treated by the codec implementation. Contains basic information, such as sample rate and number of channels,
api/audio_codecs/audio_format.h:77
↓ 10 callersClassEncodedInfo
api/audio_codecs/audio_encoder.h:106
↓ 10 callersClassFakeEncoding
test/mock_audio_encoder.h:62
↓ 10 callersClassRange
modules/desktop_capture/desktop_region_unittest.cc:671
↓ 10 callersClassRtpSource
api/transport/rtp/rtp_source.h:27
↓ 10 callersClassSimulateOutgoingTrafficIn
modules/congestion_controller/goog_cc/alr_detector_unittest.cc:27
↓ 10 callersClassSource
api/audio/audio_mixer.h:27
↓ 10 callersClassStats
call/call.h:41
↓ 10 callersClassVideoBitrateAllocation
call/rtp_video_sender_interface.h:28
↓ 9 callersClassCall
media/base/media_engine.h:33
↓ 9 callersClassConfig
modules/audio_processing/test/conversational_speech/config.h:20
↓ 9 callersClassDefaultNetEqFactory
modules/audio_coding/neteq/default_neteq_factory.h:24
↓ 9 callersClassEchoPathVariability
modules/audio_processing/aec3/echo_path_variability.h:16
↓ 9 callersClassEmpty
Empty class for use in libjingle_peerconnection_java because all targets require at least one Java file.
sdk/android/src/java/org/webrtc/Empty.java:17
↓ 9 callersClassFilter
api/audio/echo_canceller3_config.h:63
↓ 9 callersClassRtpHeaderExtensionMap
modules/rtp_rtcp/include/rtp_header_extension_map.h:26
↓ 9 callersClassSample
modules/congestion_controller/bbr/windowed_filter.h:153
↓ 9 callersClassThresholdCurve
modules/audio_coding/audio_network_adaptor/util/threshold_curve.h:18
↓ 8 callersClassBitPattern
Class for matching bit patterns such as "x1xx0000" where 'x' is allowed to be either 0 or 1.
media/base/h264_profile_level_id.cc:44
↓ 8 callersClassDesktopRect
modules/desktop_capture/win/screen_capturer_win_magnifier.h:30
↓ 8 callersClassEchoCanceller3Tester
modules/audio_processing/aec3/echo_canceller3_unittest.cc:166
↓ 8 callersClassIceParameters
p2p/base/transport_description.h:60
↓ 8 callersClassIterable
Provides a convenient way to iterate over a Java Iterable using the C++ range-for loop. E.g. for (jobject value : Iterable(jni, j_iterable)) { ... } N
sdk/android/native_api/jni/java_types.h:51
↓ 8 callersClassLocalAndRemoteSdp
test/pc/e2e/sdp/sdp_changer.h:50
↓ 8 callersClassMediaConfig
media/engine/webrtc_video_engine.h:43
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