All Classes and Interfaces
Class
Description
Audio format converter to remix and resample audio input data.
Audio options to control AudioTracks.
AudioProcessing provides a collection of voice processing components designed
for real-time communications software.
This config is intended to be used during setup of
AudioProcessing to
enable/disable audio processing effects.Audio sampling rate converter.
A source for one or more AudioTracks.
Callback interface used to obtain
RTCSessionDescriptions when calling
either createOffer or createAnswer.A device change listener is notified whenever a media device such as a
camera, microphone, or speaker is connected to or removed from the system.
"Four character code" (4CC) enumeration mainly used for conversion.
Sink interface that receives WebRTC log messages.
Base class for sources.
A MediaStream is used to group several MediaStreamTrack objects into one unit
that can be recorded or rendered.
The MediaStreamTrack represents media of a single type that originates from
one media source, e.g.
This listener gets notified whenever a track, on which this listener is
registered, ends.
A 'mute' condition change listener is notified whenever a media track has
been (un)muted.
This class wraps the native WebRTC I420BufferInterface.
The PeerConnectionFactory is the main entry point for a WebRTC application.
RTCPeerConnection callback interface.
Power management assertions to wake the display and prevent it from going to sleep on user idle.
The RTCAnswerOptions describe options specific to the session description of
type answer.
The bundle policy affects which media tracks are negotiated if the remote
endpoint is not bundle-aware, and what ICE candidates are gathered.
Represents a certificate used to authenticate WebRTC communications.
The RTCConfiguration defines a set of parameters to configure how the
peer-to-peer communication established via
RTCPeerConnection is
established or re-established.Represents a bi-directional data channel between two peers.
A NIO based buffer used to send data over an
RTCDataChannel.The RTCDataChannelInit describes options to configure a
RTCDataChannel.Used to receive events from the
RTCDataChannel.Represents the state of the
RTCDataChannel.Allows an application to access information about the Datagram Transport
Layer Security (DTLS) transport over which RTP and RTCP packets are sent and
received by RTCRtpSender's and RTCRtpReceiver's, as well other data such as
SCTP packets sent and received by data channels.
Receives events from an
RTCDtlsTransport.Datagram Transport Layer Security (DTLS) transport states.
Represents the signaling DTX status.
An ICE candidate describes the protocols and routing needed to be able to
communicate with a remote device.
Describes the current state of the ICE agent and its connection to the ICE
server (that is, the STUN or TURN server).
Describes the ICE gathering state for the
RTCPeerConnection.The RTCIceServer is used to describe the STUN and TURN servers that can be
used by the ICE Agent to establish a connection with a peer.
Allows an application access to information about the ICE transport over
which packets are sent and received.
Defines the ICE candidate policy the application uses to surface the
permitted candidates to the application; only these candidates will be used
for connectivity checks.
The RTCOfferAnswerOptions describe the options that can be used to control
the offer/answer creation process.
The RTCAnswerOptions describe options specific to the session description of
type answer.
The RTCPeerConnection represents a WebRTC connection between the local
computer and a remote peer.
This event occurs when the
RTCPeerConnection fails to gather an ICE
candidate.Indicates the current state of the
RTCPeerConnection.Describes the priority of media and data flows.
The RtcpMuxPolicy affects what ICE candidates are gathered to support
non-multiplexed RTCP.
Provides information on RTCP settings.
Contains the capabilities of the system for receiving media.
Represents the static capabilities of an endpoint's implementation of a
codec.
Describes the configuration parameters for a single media codec.
Contains information about a given contributing source (CSRC).
Describes encoding options of an
RTCRtpSender.Describes supported RTP header extensions.
Enables an application to determine whether a header extension is configured
for use within an RTCRtpSender or RTCRtpReceiver.
Describes RTP stack settings used by both
RTCRtpSenders and RTCRtpReceivers.The RTCRtpReceiver allows an application to inspect the receipt of a
MediaStreamTrack.The RTCRtpSender allows an application to control how a given
MediaStreamTrack is encoded and transmitted to a remote peer.Specifies RTP and RTCP parameters for an
RTCRtpSender.Contains information about a given synchronization source (SSRC).
Represents a combination of an
RTCRtpSender and an RTCRtpReceiver that share a common media ID (mid).Indicates the RTCRtpTransceiver's preferred directionality.
Provides configuration options for
RTCRtpTransceivers.The RTCSdpType describes the type of an
RTCSessionDescription.The RTCSessionDescription class is used by an RTCPeerConnection to expose
local and remote session descriptions.
Describes the state of the signaling process on the local end of the
RTCPeerConnection.
RTCStats represents the stats object constructed by inspecting a specific
monitored object at a specific moment in time.
An RTCStatsCollectorCallback reports back when an
RTCStatsReport is
ready.Each RTCStatsReport contains multiple RTCStats objects; one for each
underlying object (codec, stream, transport, etc.) that was inspected to
produce the stats.
An RTCStatsType indicates the type of the object that the
RTCStats
object represents.Callback interface used to get notified when the
RTCPeerConnection
has successfully set an local or remote RTCSessionDescription by
calling either setLocalDescription or setRemoteDescription.TLS certificate policy.
Base class for frame buffers of different types of pixel format and storage.
A source for one or more VideoTracks.