tvi.webrtc

Class PeerConnection

java.lang.Object

tvi.webrtc.PeerConnection

public class PeerConnection
extends java.lang.Object

Java-land version of the PeerConnection APIs; wraps the C++ API http://www.webrtc.org/reference/native-apis, which in turn is inspired by the JS APIs: http://dev.w3.org/2011/webrtc/editor/webrtc.html and http://www.w3.org/TR/mediacapture-streams/

Nested Class Summary

Nested Classes

Modifier and Type	Class and Description
static class	PeerConnection.AdapterType
static class	PeerConnection.BundlePolicy Java version of PeerConnectionInterface.BundlePolicy
static class	PeerConnection.CandidateNetworkPolicy Java version of PeerConnectionInterface.CandidateNetworkPolicy
static class	PeerConnection.ContinualGatheringPolicy Java version of PeerConnectionInterface.ContinualGatheringPolicy
static class	PeerConnection.IceConnectionState Tracks PeerConnectionInterface::IceConnectionState
static class	PeerConnection.IceGatheringState Tracks PeerConnectionInterface::IceGatheringState
static class	PeerConnection.IceServer Java version of PeerConnectionInterface.IceServer.
static class	PeerConnection.IceTransportsType Java version of PeerConnectionInterface.IceTransportsType
static class	PeerConnection.KeyType Java version of rtc::KeyType
static interface	PeerConnection.Observer Java version of PeerConnectionObserver.
static class	PeerConnection.PeerConnectionState Tracks PeerConnectionInterface::PeerConnectionState
static class	PeerConnection.PortPrunePolicy Java version of webrtc::PortPrunePolicy
static class	PeerConnection.RTCConfiguration Java version of PeerConnectionInterface.RTCConfiguration
static class	PeerConnection.RtcpMuxPolicy Java version of PeerConnectionInterface.RtcpMuxPolicy
static class	PeerConnection.SdpSemantics Java version of webrtc::SdpSemantics.
static class	PeerConnection.SignalingState Tracks PeerConnectionInterface::SignalingState
static class	PeerConnection.TcpCandidatePolicy Java version of PeerConnectionInterface.TcpCandidatePolicy
static class	PeerConnection.TlsCertPolicy Tracks PeerConnectionInterface::TlsCertPolicy

Constructor Summary

Constructors

Constructor and Description
PeerConnection(NativePeerConnectionFactory factory) Wraps a PeerConnection created by the factory.

Method Summary

Modifier and Type	Method and Description
boolean	addIceCandidate(IceCandidate candidate)
boolean	addStream(MediaStream stream) Adds a new MediaStream to be sent on this peer connection.
RtpSender	addTrack(MediaStreamTrack track) Adds a new media stream track to be sent on this peer connection, and returns the newly created RtpSender.
RtpSender	addTrack(MediaStreamTrack track, java.util.List<java.lang.String> streamIds)
RtpTransceiver	addTransceiver(MediaStreamTrack.MediaType mediaType)
RtpTransceiver	addTransceiver(MediaStreamTrack.MediaType mediaType, RtpTransceiver.RtpTransceiverInit init)
RtpTransceiver	addTransceiver(MediaStreamTrack track) Creates a new RtpTransceiver and adds it to the set of transceivers.
RtpTransceiver	addTransceiver(MediaStreamTrack track, RtpTransceiver.RtpTransceiverInit init)
void	close()
PeerConnection.PeerConnectionState	connectionState()
void	createAnswer(SdpObserver observer, MediaConstraints constraints)
DataChannel	createDataChannel(java.lang.String label, DataChannel.Init init)
static long	createNativePeerConnectionObserver(PeerConnection.Observer observer)
void	createOffer(SdpObserver observer, MediaConstraints constraints)
RtpSender	createSender(java.lang.String kind, java.lang.String stream_id) Creates an RtpSender without a track.
void	dispose() Free native resources associated with this PeerConnection instance.
RtcCertificatePem	getCertificate()
SessionDescription	getLocalDescription()
long	getNativePeerConnection() Returns a pointer to the native webrtc::PeerConnectionInterface.
java.util.List<RtpReceiver>	getReceivers() Gets all RtpReceivers associated with this peer connection.
SessionDescription	getRemoteDescription()
java.util.List<RtpSender>	getSenders() Gets all RtpSenders associated with this peer connection.
void	getStats(RTCStatsCollectorCallback callback) Gets stats using the new stats collection API, see webrtc/api/stats/.
boolean	getStats(StatsObserver observer, MediaStreamTrack track) Deprecated.
java.util.List<RtpTransceiver>	getTransceivers() Gets all RtpTransceivers associated with this peer connection.
PeerConnection.IceConnectionState	iceConnectionState()
PeerConnection.IceGatheringState	iceGatheringState()
boolean	removeIceCandidates(IceCandidate[] candidates)
void	removeStream(MediaStream stream) Removes the given media stream from this peer connection.
boolean	removeTrack(RtpSender sender) Stops sending media from sender.
void	setAudioPlayout(boolean playout) Enables/disables playout of received audio streams.
void	setAudioRecording(boolean recording) Enables/disables recording of transmitted audio streams.
boolean	setBitrate(java.lang.Integer min, java.lang.Integer current, java.lang.Integer max) Limits the bandwidth allocated for all RTP streams sent by this PeerConnection.
boolean	setConfiguration(PeerConnection.RTCConfiguration config)
void	setLocalDescription(SdpObserver observer, SessionDescription sdp)
void	setRemoteDescription(SdpObserver observer, SessionDescription sdp)
PeerConnection.SignalingState	signalingState()
boolean	startRtcEventLog(int file_descriptor, int max_size_bytes) Starts recording an RTC event log.
void	stopRtcEventLog() Stops recording an RTC event log.

Methods inherited from class java.lang.Object

clone, equals, finalize, getClass, hashCode, notify, notifyAll, toString, wait, wait, wait

Constructor Detail

PeerConnection

public PeerConnection(NativePeerConnectionFactory factory)

Wraps a PeerConnection created by the factory. Can be used by clients that want to implement their PeerConnection creation in JNI.

Method Detail

getLocalDescription

public SessionDescription getLocalDescription()

getRemoteDescription

public SessionDescription getRemoteDescription()

getCertificate

public RtcCertificatePem getCertificate()

createDataChannel

public DataChannel createDataChannel(java.lang.String label,
                                     DataChannel.Init init)

createOffer

public void createOffer(SdpObserver observer,
                        MediaConstraints constraints)

createAnswer

public void createAnswer(SdpObserver observer,
                         MediaConstraints constraints)

setLocalDescription

public void setLocalDescription(SdpObserver observer,
                                SessionDescription sdp)

setRemoteDescription

public void setRemoteDescription(SdpObserver observer,
                                 SessionDescription sdp)

setAudioPlayout

public void setAudioPlayout(boolean playout)

Enables/disables playout of received audio streams. Enabled by default. Note that even if playout is enabled, streams will only be played out if the appropriate SDP is also applied. The main purpose of this API is to be able to control the exact time when audio playout starts.

setAudioRecording

public void setAudioRecording(boolean recording)

Enables/disables recording of transmitted audio streams. Enabled by default. Note that even if recording is enabled, streams will only be recorded if the appropriate SDP is also applied. The main purpose of this API is to be able to control the exact time when audio recording starts.

setConfiguration

public boolean setConfiguration(PeerConnection.RTCConfiguration config)

addIceCandidate

public boolean addIceCandidate(IceCandidate candidate)

removeIceCandidates

public boolean removeIceCandidates(IceCandidate[] candidates)

addStream

public boolean addStream(MediaStream stream)

Adds a new MediaStream to be sent on this peer connection. Note: This method is not supported with SdpSemantics.UNIFIED_PLAN. Please use addTrack instead.

removeStream

public void removeStream(MediaStream stream)

Removes the given media stream from this peer connection. This method is not supported with SdpSemantics.UNIFIED_PLAN. Please use removeTrack instead.

createSender

public RtpSender createSender(java.lang.String kind,
                              java.lang.String stream_id)

Creates an RtpSender without a track.

This method allows an application to cause the PeerConnection to negotiate sending/receiving a specific media type, but without having a track to send yet.

When the application does want to begin sending a track, it can call RtpSender.setTrack, which doesn't require any additional SDP negotiation.

Example use:

 
 audioSender = pc.createSender("audio", "stream1");
 videoSender = pc.createSender("video", "stream1");
 // Do normal SDP offer/answer, which will kick off ICE/DTLS and negotiate
 // media parameters....
 // Later, when the endpoint is ready to actually begin sending:
 audioSender.setTrack(audioTrack, false);
 videoSender.setTrack(videoTrack, false);
 
 

Note: This corresponds most closely to "addTransceiver" in the official WebRTC API, in that it creates a sender without a track. It was implemented before addTransceiver because it provides useful functionality, and properly implementing transceivers would have required a great deal more work.

Note: This is only available with SdpSemantics.PLAN_B specified. Please use addTransceiver instead.

Parameters:

kind - Corresponds to MediaStreamTrack kinds (must be "audio" or "video").

stream_id - The ID of the MediaStream that this sender's track will be associated with when SDP is applied to the remote PeerConnection. If createSender is used to create an audio and video sender that should be synchronized, they should use the same stream ID.

Returns:

A new RtpSender object if successful, or null otherwise.

getSenders

public java.util.List<RtpSender> getSenders()

Gets all RtpSenders associated with this peer connection. Note that calling getSenders will dispose of the senders previously returned.

getReceivers

public java.util.List<RtpReceiver> getReceivers()

Gets all RtpReceivers associated with this peer connection. Note that calling getReceivers will dispose of the receivers previously returned.

getTransceivers

public java.util.List<RtpTransceiver> getTransceivers()

Gets all RtpTransceivers associated with this peer connection. Note that calling getTransceivers will dispose of the transceivers previously returned. Note: This is only available with SdpSemantics.UNIFIED_PLAN specified.

addTrack

public RtpSender addTrack(MediaStreamTrack track)

Adds a new media stream track to be sent on this peer connection, and returns the newly created RtpSender. If streamIds are specified, the RtpSender will be associated with the streams specified in the streamIds list.

Throws:

java.lang.IllegalStateException - if an error accors in C++ addTrack. An error can occur if: - A sender already exists for the track. - The peer connection is closed.

addTrack

public RtpSender addTrack(MediaStreamTrack track,
                          java.util.List<java.lang.String> streamIds)

removeTrack

public boolean removeTrack(RtpSender sender)

Stops sending media from sender. The sender will still appear in getSenders. Future calls to createOffer will mark the m section for the corresponding transceiver as receive only or inactive, as defined in JSEP. Returns true on success.

addTransceiver

public RtpTransceiver addTransceiver(MediaStreamTrack track)

Creates a new RtpTransceiver and adds it to the set of transceivers. Adding a transceiver will cause future calls to CreateOffer to add a media description for the corresponding transceiver.

The initial value of |mid| in the returned transceiver is null. Setting a new session description may change it to a non-null value.

https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver

If a MediaStreamTrack is specified then a transceiver will be added with a sender set to transmit the given track. The kind of the transceiver (and sender/receiver) will be derived from the kind of the track.

If MediaType is specified then a transceiver will be added based upon that type. This can be either MEDIA_TYPE_AUDIO or MEDIA_TYPE_VIDEO.

Optionally, an RtpTransceiverInit structure can be specified to configure the transceiver from construction. If not specified, the transceiver will default to having a direction of kSendRecv and not be part of any streams.

Note: These methods are only available with SdpSemantics.UNIFIED_PLAN specified.

Throws:

java.lang.IllegalStateException - if an error accors in C++ addTransceiver

addTransceiver

public RtpTransceiver addTransceiver(MediaStreamTrack track,
                                     @Nullable
                                     RtpTransceiver.RtpTransceiverInit init)

addTransceiver

public RtpTransceiver addTransceiver(MediaStreamTrack.MediaType mediaType)

addTransceiver

public RtpTransceiver addTransceiver(MediaStreamTrack.MediaType mediaType,
                                     @Nullable
                                     RtpTransceiver.RtpTransceiverInit init)

getStats

@Deprecated
public boolean getStats(StatsObserver observer,
                                    @Nullable
                                    MediaStreamTrack track)

Deprecated.

getStats

public void getStats(RTCStatsCollectorCallback callback)

Gets stats using the new stats collection API, see webrtc/api/stats/. These will replace old stats collection API when the new API has matured enough.

setBitrate

public boolean setBitrate(java.lang.Integer min,
                          java.lang.Integer current,
                          java.lang.Integer max)

Limits the bandwidth allocated for all RTP streams sent by this PeerConnection. Pass null to leave a value unchanged.

startRtcEventLog

public boolean startRtcEventLog(int file_descriptor,
                                int max_size_bytes)

Starts recording an RTC event log. Ownership of the file is transfered to the native code. If an RTC event log is already being recorded, it will be stopped and a new one will start using the provided file. Logging will continue until the stopRtcEventLog function is called. The max_size_bytes argument is ignored, it is added for future use.

stopRtcEventLog

public void stopRtcEventLog()

Stops recording an RTC event log. If no RTC event log is currently being recorded, this call will have no effect.

signalingState

public PeerConnection.SignalingState signalingState()

iceConnectionState

public PeerConnection.IceConnectionState iceConnectionState()

connectionState

public PeerConnection.PeerConnectionState connectionState()

iceGatheringState

public PeerConnection.IceGatheringState iceGatheringState()

close

public void close()

dispose

public void dispose()

Free native resources associated with this PeerConnection instance. This method removes a reference count from the C++ PeerConnection object, which should result in it being destroyed. It also calls equivalent "dispose" methods on the Java objects attached to this PeerConnection (streams, senders, receivers), such that their associated C++ objects will also be destroyed.

Note that this method cannot be safely called from an observer callback (PeerConnection.Observer, DataChannel.Observer, etc.). If you want to, for example, destroy the PeerConnection after an "ICE failed" callback, you must do this asynchronously (in other words, unwind the stack first). See bug 3721 for more details.

getNativePeerConnection

public long getNativePeerConnection()

Returns a pointer to the native webrtc::PeerConnectionInterface.

createNativePeerConnectionObserver

public static long createNativePeerConnectionObserver(PeerConnection.Observer observer)