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/*
* Copyright 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
package org.webrtc;
import java.util.ArrayList;
import java.util.Arrays;
import java.util.Collections;
import java.util.HashMap;
import java.util.List;
import java.util.Map;
import javax.annotation.Nullable;
/**
* 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/
*/
public class PeerConnection {
/** Tracks PeerConnectionInterface::IceGatheringState */
public enum IceGatheringState {
NEW,
GATHERING,
COMPLETE;
@CalledByNative("IceGatheringState")
static IceGatheringState fromNativeIndex(int nativeIndex) {
return values()[nativeIndex];
}
}
/** Tracks PeerConnectionInterface::IceConnectionState */
public enum IceConnectionState {
NEW,
CHECKING,
CONNECTED,
COMPLETED,
FAILED,
DISCONNECTED,
CLOSED;
@CalledByNative("IceConnectionState")
static IceConnectionState fromNativeIndex(int nativeIndex) {
return values()[nativeIndex];
}
}
/** Tracks PeerConnectionInterface::PeerConnectionState */
public enum PeerConnectionState {
NEW,
CONNECTING,
CONNECTED,
DISCONNECTED,
FAILED,
CLOSED;
@CalledByNative("PeerConnectionState")
static PeerConnectionState fromNativeIndex(int nativeIndex) {
return values()[nativeIndex];
}
}
/** Tracks PeerConnectionInterface::TlsCertPolicy */
public enum TlsCertPolicy {
TLS_CERT_POLICY_SECURE,
TLS_CERT_POLICY_INSECURE_NO_CHECK,
}
/** Tracks PeerConnectionInterface::SignalingState */
public enum SignalingState {
STABLE,
HAVE_LOCAL_OFFER,
HAVE_LOCAL_PRANSWER,
HAVE_REMOTE_OFFER,
HAVE_REMOTE_PRANSWER,
CLOSED;
@CalledByNative("SignalingState")
static SignalingState fromNativeIndex(int nativeIndex) {
return values()[nativeIndex];
}
}
/** Java version of PeerConnectionObserver. */
public static interface Observer {
/** Triggered when the SignalingState changes. */
@CalledByNative("Observer") void onSignalingChange(SignalingState newState);
/** Triggered when the IceConnectionState changes. */
@CalledByNative("Observer") void onIceConnectionChange(IceConnectionState newState);
/* Triggered when the standard-compliant state transition of IceConnectionState happens. */
@CalledByNative("Observer")
default void onStandardizedIceConnectionChange(IceConnectionState newState) {}
/** Triggered when the PeerConnectionState changes. */
@CalledByNative("Observer")
default void onConnectionChange(PeerConnectionState newState) {}
/** Triggered when the ICE connection receiving status changes. */
@CalledByNative("Observer") void onIceConnectionReceivingChange(boolean receiving);
/** Triggered when the IceGatheringState changes. */
@CalledByNative("Observer") void onIceGatheringChange(IceGatheringState newState);
/** Triggered when a new ICE candidate has been found. */
@CalledByNative("Observer") void onIceCandidate(IceCandidate candidate);
/** Triggered when some ICE candidates have been removed. */
@CalledByNative("Observer") void onIceCandidatesRemoved(IceCandidate[] candidates);
/** Triggered when the ICE candidate pair is changed. */
@CalledByNative("Observer")
default void onSelectedCandidatePairChanged(CandidatePairChangeEvent event) {}
/** Triggered when media is received on a new stream from remote peer. */
@CalledByNative("Observer") void onAddStream(MediaStream stream);
/** Triggered when a remote peer close a stream. */
@CalledByNative("Observer") void onRemoveStream(MediaStream stream);
/** Triggered when a remote peer opens a DataChannel. */
@CalledByNative("Observer") void onDataChannel(DataChannel dataChannel);
/** Triggered when renegotiation is necessary. */
@CalledByNative("Observer") void onRenegotiationNeeded();
/**
* Triggered when a new track is signaled by the remote peer, as a result of
* setRemoteDescription.
*/
@CalledByNative("Observer") void onAddTrack(RtpReceiver receiver, MediaStream[] mediaStreams);
/**
* Triggered when the signaling from SetRemoteDescription indicates that a transceiver
* will be receiving media from a remote endpoint. This is only called if UNIFIED_PLAN
* semantics are specified. The transceiver will be disposed automatically.
*/
@CalledByNative("Observer") default void onTrack(RtpTransceiver transceiver){};
}
/** Java version of PeerConnectionInterface.IceServer. */
public static class IceServer {
// List of URIs associated with this server. Valid formats are described
// in RFC7064 and RFC7065, and more may be added in the future. The "host"
// part of the URI may contain either an IP address or a hostname.
@Deprecated public final String uri;
public final List urls;
public final String username;
public final String password;
public final TlsCertPolicy tlsCertPolicy;
// If the URIs in |urls| only contain IP addresses, this field can be used
// to indicate the hostname, which may be necessary for TLS (using the SNI
// extension). If |urls| itself contains the hostname, this isn't
// necessary.
public final String hostname;
// List of protocols to be used in the TLS ALPN extension.
public final List tlsAlpnProtocols;
// List of elliptic curves to be used in the TLS elliptic curves extension.
// Only curve names supported by OpenSSL should be used (eg. "P-256","X25519").
public final List tlsEllipticCurves;
/** Convenience constructor for STUN servers. */
@Deprecated
public IceServer(String uri) {
this(uri, "", "");
}
@Deprecated
public IceServer(String uri, String username, String password) {
this(uri, username, password, TlsCertPolicy.TLS_CERT_POLICY_SECURE);
}
@Deprecated
public IceServer(String uri, String username, String password, TlsCertPolicy tlsCertPolicy) {
this(uri, username, password, tlsCertPolicy, "");
}
@Deprecated
public IceServer(String uri, String username, String password, TlsCertPolicy tlsCertPolicy,
String hostname) {
this(uri, Collections.singletonList(uri), username, password, tlsCertPolicy, hostname, null,
null);
}
private IceServer(String uri, List urls, String username, String password,
TlsCertPolicy tlsCertPolicy, String hostname, List tlsAlpnProtocols,
List tlsEllipticCurves) {
if (uri == null || urls == null || urls.isEmpty()) {
throw new IllegalArgumentException("uri == null || urls == null || urls.isEmpty()");
}
for (String it : urls) {
if (it == null) {
throw new IllegalArgumentException("urls element is null: " + urls);
}
}
if (username == null) {
throw new IllegalArgumentException("username == null");
}
if (password == null) {
throw new IllegalArgumentException("password == null");
}
if (hostname == null) {
throw new IllegalArgumentException("hostname == null");
}
this.uri = uri;
this.urls = urls;
this.username = username;
this.password = password;
this.tlsCertPolicy = tlsCertPolicy;
this.hostname = hostname;
this.tlsAlpnProtocols = tlsAlpnProtocols;
this.tlsEllipticCurves = tlsEllipticCurves;
}
@Override
public String toString() {
return urls + " [" + username + ":" + password + "] [" + tlsCertPolicy + "] [" + hostname
+ "] [" + tlsAlpnProtocols + "] [" + tlsEllipticCurves + "]";
}
@Override
public boolean equals(@Nullable Object obj) {
if (obj == null) {
return false;
}
if (obj == this) {
return true;
}
if (!(obj instanceof IceServer)) {
return false;
}
IceServer other = (IceServer) obj;
return (uri.equals(other.uri) && urls.equals(other.urls) && username.equals(other.username)
&& password.equals(other.password) && tlsCertPolicy.equals(other.tlsCertPolicy)
&& hostname.equals(other.hostname) && tlsAlpnProtocols.equals(other.tlsAlpnProtocols)
&& tlsEllipticCurves.equals(other.tlsEllipticCurves));
}
@Override
public int hashCode() {
Object[] values = {uri, urls, username, password, tlsCertPolicy, hostname, tlsAlpnProtocols,
tlsEllipticCurves};
return Arrays.hashCode(values);
}
public static Builder builder(String uri) {
return new Builder(Collections.singletonList(uri));
}
public static Builder builder(List urls) {
return new Builder(urls);
}
public static class Builder {
@Nullable private final List urls;
private String username = "";
private String password = "";
private TlsCertPolicy tlsCertPolicy = TlsCertPolicy.TLS_CERT_POLICY_SECURE;
private String hostname = "";
private List tlsAlpnProtocols;
private List tlsEllipticCurves;
private Builder(List urls) {
if (urls == null || urls.isEmpty()) {
throw new IllegalArgumentException("urls == null || urls.isEmpty(): " + urls);
}
this.urls = urls;
}
public Builder setUsername(String username) {
this.username = username;
return this;
}
public Builder setPassword(String password) {
this.password = password;
return this;
}
public Builder setTlsCertPolicy(TlsCertPolicy tlsCertPolicy) {
this.tlsCertPolicy = tlsCertPolicy;
return this;
}
public Builder setHostname(String hostname) {
this.hostname = hostname;
return this;
}
public Builder setTlsAlpnProtocols(List tlsAlpnProtocols) {
this.tlsAlpnProtocols = tlsAlpnProtocols;
return this;
}
public Builder setTlsEllipticCurves(List tlsEllipticCurves) {
this.tlsEllipticCurves = tlsEllipticCurves;
return this;
}
public IceServer createIceServer() {
return new IceServer(urls.get(0), urls, username, password, tlsCertPolicy, hostname,
tlsAlpnProtocols, tlsEllipticCurves);
}
}
@Nullable
@CalledByNative("IceServer")
List getUrls() {
return urls;
}
@Nullable
@CalledByNative("IceServer")
String getUsername() {
return username;
}
@Nullable
@CalledByNative("IceServer")
String getPassword() {
return password;
}
@CalledByNative("IceServer")
TlsCertPolicy getTlsCertPolicy() {
return tlsCertPolicy;
}
@Nullable
@CalledByNative("IceServer")
String getHostname() {
return hostname;
}
@CalledByNative("IceServer")
List getTlsAlpnProtocols() {
return tlsAlpnProtocols;
}
@CalledByNative("IceServer")
List getTlsEllipticCurves() {
return tlsEllipticCurves;
}
}
/** Java version of PeerConnectionInterface.IceTransportsType */
public enum IceTransportsType { NONE, RELAY, NOHOST, ALL }
/** Java version of PeerConnectionInterface.BundlePolicy */
public enum BundlePolicy { BALANCED, MAXBUNDLE, MAXCOMPAT }
/** Java version of PeerConnectionInterface.RtcpMuxPolicy */
public enum RtcpMuxPolicy { NEGOTIATE, REQUIRE }
/** Java version of PeerConnectionInterface.TcpCandidatePolicy */
public enum TcpCandidatePolicy { ENABLED, DISABLED }
/** Java version of PeerConnectionInterface.CandidateNetworkPolicy */
public enum CandidateNetworkPolicy { ALL, LOW_COST }
// Keep in sync with webrtc/rtc_base/network_constants.h.
public enum AdapterType {
UNKNOWN(0),
ETHERNET(1 << 0),
WIFI(1 << 1),
CELLULAR(1 << 2),
VPN(1 << 3),
LOOPBACK(1 << 4),
ADAPTER_TYPE_ANY(1 << 5);
public final Integer bitMask;
private AdapterType(Integer bitMask) {
this.bitMask = bitMask;
}
private static final Map BY_BITMASK = new HashMap<>();
static {
for (AdapterType t : values()) {
BY_BITMASK.put(t.bitMask, t);
}
}
@CalledByNative("AdapterType")
static AdapterType fromNativeIndex(int nativeIndex) {
return BY_BITMASK.get(nativeIndex);
}
}
/** Java version of rtc::KeyType */
public enum KeyType { RSA, ECDSA }
/** Java version of PeerConnectionInterface.ContinualGatheringPolicy */
public enum ContinualGatheringPolicy { GATHER_ONCE, GATHER_CONTINUALLY }
/** Java version of webrtc::PortPrunePolicy */
public enum PortPrunePolicy {
NO_PRUNE, // Do not prune turn port.
PRUNE_BASED_ON_PRIORITY, // Prune turn port based the priority on the same network
KEEP_FIRST_READY // Keep the first ready port and prune the rest on the same network.
}
/** Java version of rtc::IntervalRange */
public static class IntervalRange {
private final int min;
private final int max;
public IntervalRange(int min, int max) {
this.min = min;
this.max = max;
}
@CalledByNative("IntervalRange")
public int getMin() {
return min;
}
@CalledByNative("IntervalRange")
public int getMax() {
return max;
}
}
/**
* Java version of webrtc::SdpSemantics.
*
* Configure the SDP semantics used by this PeerConnection. Note that the
* WebRTC 1.0 specification requires UNIFIED_PLAN semantics. The
* RtpTransceiver API is only available with UNIFIED_PLAN semantics.
*
* PLAN_B will cause PeerConnection to create offers and answers with at
* most one audio and one video m= section with multiple RtpSenders and
* RtpReceivers specified as multiple a=ssrc lines within the section. This
* will also cause PeerConnection to ignore all but the first m= section of
* the same media type.
*
*
UNIFIED_PLAN will cause PeerConnection to create offers and answers with
* multiple m= sections where each m= section maps to one RtpSender and one
* RtpReceiver (an RtpTransceiver), either both audio or both video. This
* will also cause PeerConnection to ignore all but the first a=ssrc lines
* that form a Plan B stream.
*
*
For users who wish to send multiple audio/video streams and need to stay
* interoperable with legacy WebRTC implementations, specify PLAN_B.
*
*
For users who wish to send multiple audio/video streams and/or wish to
* use the new RtpTransceiver API, specify UNIFIED_PLAN.
*/
public enum SdpSemantics { PLAN_B, UNIFIED_PLAN }
/** Java version of PeerConnectionInterface.RTCConfiguration */
// TODO(qingsi): Resolve the naming inconsistency of fields with/without units.
public static class RTCConfiguration {
public IceTransportsType iceTransportsType;
public List iceServers;
public BundlePolicy bundlePolicy;
@Nullable public RtcCertificatePem certificate;
public RtcpMuxPolicy rtcpMuxPolicy;
public TcpCandidatePolicy tcpCandidatePolicy;
public CandidateNetworkPolicy candidateNetworkPolicy;
public int audioJitterBufferMaxPackets;
public boolean audioJitterBufferFastAccelerate;
public int iceConnectionReceivingTimeout;
public int iceBackupCandidatePairPingInterval;
public KeyType keyType;
public ContinualGatheringPolicy continualGatheringPolicy;
public int iceCandidatePoolSize;
@Deprecated // by the turnPortPrunePolicy. See bugs.webrtc.org/11026
public boolean pruneTurnPorts;
public PortPrunePolicy turnPortPrunePolicy;
public boolean presumeWritableWhenFullyRelayed;
public boolean surfaceIceCandidatesOnIceTransportTypeChanged;
// The following fields define intervals in milliseconds at which ICE
// connectivity checks are sent.
//
// We consider ICE is "strongly connected" for an agent when there is at
// least one candidate pair that currently succeeds in connectivity check
// from its direction i.e. sending a ping and receives a ping response, AND
// all candidate pairs have sent a minimum number of pings for connectivity
// (this number is implementation-specific). Otherwise, ICE is considered in
// "weak connectivity".
//
// Note that the above notion of strong and weak connectivity is not defined
// in RFC 5245, and they apply to our current ICE implementation only.
//
// 1) iceCheckIntervalStrongConnectivityMs defines the interval applied to
// ALL candidate pairs when ICE is strongly connected,
// 2) iceCheckIntervalWeakConnectivityMs defines the counterpart for ALL
// pairs when ICE is weakly connected, and
// 3) iceCheckMinInterval defines the minimal interval (equivalently the
// maximum rate) that overrides the above two intervals when either of them
// is less.
@Nullable public Integer iceCheckIntervalStrongConnectivityMs;
@Nullable public Integer iceCheckIntervalWeakConnectivityMs;
@Nullable public Integer iceCheckMinInterval;
// The time period in milliseconds for which a candidate pair must wait for response to
// connectivitiy checks before it becomes unwritable.
@Nullable public Integer iceUnwritableTimeMs;
// The minimum number of connectivity checks that a candidate pair must sent without receiving
// response before it becomes unwritable.
@Nullable public Integer iceUnwritableMinChecks;
// The interval in milliseconds at which STUN candidates will resend STUN binding requests
// to keep NAT bindings open.
// The default value in the implementation is used if this field is null.
@Nullable public Integer stunCandidateKeepaliveIntervalMs;
public boolean disableIPv6OnWifi;
// By default, PeerConnection will use a limited number of IPv6 network
// interfaces, in order to avoid too many ICE candidate pairs being created
// and delaying ICE completion.
//
// Can be set to Integer.MAX_VALUE to effectively disable the limit.
public int maxIPv6Networks;
@Nullable public IntervalRange iceRegatherIntervalRange;
// These values will be overridden by MediaStream constraints if deprecated constraints-based
// create peerconnection interface is used.
public boolean disableIpv6;
public boolean enableDscp;
public boolean enableCpuOveruseDetection;
public boolean enableRtpDataChannel;
public boolean suspendBelowMinBitrate;
@Nullable public Integer screencastMinBitrate;
@Nullable public Boolean combinedAudioVideoBwe;
@Nullable public Boolean enableDtlsSrtp;
// Use "Unknown" to represent no preference of adapter types, not the
// preference of adapters of unknown types.
public AdapterType networkPreference;
public SdpSemantics sdpSemantics;
// This is an optional wrapper for the C++ webrtc::TurnCustomizer.
@Nullable public TurnCustomizer turnCustomizer;
// Actively reset the SRTP parameters whenever the DTLS transports underneath are reset for
// every offer/answer negotiation.This is only intended to be a workaround for crbug.com/835958
public boolean activeResetSrtpParams;
/*
* Experimental flag that enables a use of media transport. If this is true, the media transport
* factory MUST be provided to the PeerConnectionFactory.
*/
public boolean useMediaTransport;
/*
* Experimental flag that enables a use of media transport for data channels. If this is true,
* the media transport factory MUST be provided to the PeerConnectionFactory.
*/
public boolean useMediaTransportForDataChannels;
/**
* Defines advanced optional cryptographic settings related to SRTP and
* frame encryption for native WebRTC. Setting this will overwrite any
* options set through the PeerConnectionFactory (which is deprecated).
*/
@Nullable public CryptoOptions cryptoOptions;
/**
* An optional string that if set will be attached to the
* TURN_ALLOCATE_REQUEST which can be used to correlate client
* logs with backend logs
*/
@Nullable public String turnLoggingId;
public int minPort;
public int maxPort;
public static int PORTALLOCATOR_DISABLE_UDP = 0x01;
public static int PORTALLOCATOR_DISABLE_STUN = 0x02;
public static int PORTALLOCATOR_DISABLE_RELAY = 0x04;
public int portAllocatorFlags;
// TODO(deadbeef): Instead of duplicating the defaults here, we should do
// something to pick up the defaults from C++. The Objective-C equivalent
// of RTCConfiguration does that.
public RTCConfiguration(List iceServers) {
iceTransportsType = IceTransportsType.ALL;
bundlePolicy = BundlePolicy.BALANCED;
rtcpMuxPolicy = RtcpMuxPolicy.REQUIRE;
tcpCandidatePolicy = TcpCandidatePolicy.ENABLED;
candidateNetworkPolicy = CandidateNetworkPolicy.ALL;
this.iceServers = iceServers;
audioJitterBufferMaxPackets = 50;
audioJitterBufferFastAccelerate = false;
iceConnectionReceivingTimeout = -1;
iceBackupCandidatePairPingInterval = -1;
keyType = KeyType.ECDSA;
continualGatheringPolicy = ContinualGatheringPolicy.GATHER_ONCE;
iceCandidatePoolSize = 0;
pruneTurnPorts = false;
turnPortPrunePolicy = PortPrunePolicy.NO_PRUNE;
presumeWritableWhenFullyRelayed = false;
surfaceIceCandidatesOnIceTransportTypeChanged = false;
iceCheckIntervalStrongConnectivityMs = null;
iceCheckIntervalWeakConnectivityMs = null;
iceCheckMinInterval = null;
iceUnwritableTimeMs = null;
iceUnwritableMinChecks = null;
stunCandidateKeepaliveIntervalMs = null;
disableIPv6OnWifi = false;
maxIPv6Networks = 5;
iceRegatherIntervalRange = null;
disableIpv6 = false;
enableDscp = false;
enableCpuOveruseDetection = true;
enableRtpDataChannel = false;
suspendBelowMinBitrate = false;
screencastMinBitrate = null;
combinedAudioVideoBwe = null;
enableDtlsSrtp = null;
networkPreference = AdapterType.UNKNOWN;
sdpSemantics = SdpSemantics.PLAN_B;
activeResetSrtpParams = false;
useMediaTransport = false;
useMediaTransportForDataChannels = false;
cryptoOptions = null;
turnLoggingId = null;
minPort = 0;
maxPort = 0;
portAllocatorFlags = 0;
}
@CalledByNative("RTCConfiguration")
IceTransportsType getIceTransportsType() {
return iceTransportsType;
}
@CalledByNative("RTCConfiguration")
List getIceServers() {
return iceServers;
}
@CalledByNative("RTCConfiguration")
BundlePolicy getBundlePolicy() {
return bundlePolicy;
}
@CalledByNative("RTCConfiguration")
PortPrunePolicy getTurnPortPrunePolicy() {
return turnPortPrunePolicy;
}
@Nullable
@CalledByNative("RTCConfiguration")
RtcCertificatePem getCertificate() {
return certificate;
}
@CalledByNative("RTCConfiguration")
RtcpMuxPolicy getRtcpMuxPolicy() {
return rtcpMuxPolicy;
}
@CalledByNative("RTCConfiguration")
TcpCandidatePolicy getTcpCandidatePolicy() {
return tcpCandidatePolicy;
}
@CalledByNative("RTCConfiguration")
CandidateNetworkPolicy getCandidateNetworkPolicy() {
return candidateNetworkPolicy;
}
@CalledByNative("RTCConfiguration")
int getAudioJitterBufferMaxPackets() {
return audioJitterBufferMaxPackets;
}
@CalledByNative("RTCConfiguration")
boolean getAudioJitterBufferFastAccelerate() {
return audioJitterBufferFastAccelerate;
}
@CalledByNative("RTCConfiguration")
int getIceConnectionReceivingTimeout() {
return iceConnectionReceivingTimeout;
}
@CalledByNative("RTCConfiguration")
int getIceBackupCandidatePairPingInterval() {
return iceBackupCandidatePairPingInterval;
}
@CalledByNative("RTCConfiguration")
KeyType getKeyType() {
return keyType;
}
@CalledByNative("RTCConfiguration")
ContinualGatheringPolicy getContinualGatheringPolicy() {
return continualGatheringPolicy;
}
@CalledByNative("RTCConfiguration")
int getIceCandidatePoolSize() {
return iceCandidatePoolSize;
}
@CalledByNative("RTCConfiguration")
boolean getPruneTurnPorts() {
return pruneTurnPorts;
}
@CalledByNative("RTCConfiguration")
boolean getPresumeWritableWhenFullyRelayed() {
return presumeWritableWhenFullyRelayed;
}
@CalledByNative("RTCConfiguration")
boolean getSurfaceIceCandidatesOnIceTransportTypeChanged() {
return surfaceIceCandidatesOnIceTransportTypeChanged;
}
@Nullable
@CalledByNative("RTCConfiguration")
Integer getIceCheckIntervalStrongConnectivity() {
return iceCheckIntervalStrongConnectivityMs;
}
@Nullable
@CalledByNative("RTCConfiguration")
Integer getIceCheckIntervalWeakConnectivity() {
return iceCheckIntervalWeakConnectivityMs;
}
@Nullable
@CalledByNative("RTCConfiguration")
Integer getIceCheckMinInterval() {
return iceCheckMinInterval;
}
@Nullable
@CalledByNative("RTCConfiguration")
Integer getIceUnwritableTimeout() {
return iceUnwritableTimeMs;
}
@Nullable
@CalledByNative("RTCConfiguration")
Integer getIceUnwritableMinChecks() {
return iceUnwritableMinChecks;
}
@Nullable
@CalledByNative("RTCConfiguration")
Integer getStunCandidateKeepaliveInterval() {
return stunCandidateKeepaliveIntervalMs;
}
@CalledByNative("RTCConfiguration")
boolean getDisableIPv6OnWifi() {
return disableIPv6OnWifi;
}
@CalledByNative("RTCConfiguration")
int getMaxIPv6Networks() {
return maxIPv6Networks;
}
@Nullable
@CalledByNative("RTCConfiguration")
IntervalRange getIceRegatherIntervalRange() {
return iceRegatherIntervalRange;
}
@Nullable
@CalledByNative("RTCConfiguration")
TurnCustomizer getTurnCustomizer() {
return turnCustomizer;
}
@CalledByNative("RTCConfiguration")
boolean getDisableIpv6() {
return disableIpv6;
}
@CalledByNative("RTCConfiguration")
boolean getEnableDscp() {
return enableDscp;
}
@CalledByNative("RTCConfiguration")
boolean getEnableCpuOveruseDetection() {
return enableCpuOveruseDetection;
}
@CalledByNative("RTCConfiguration")
boolean getEnableRtpDataChannel() {
return enableRtpDataChannel;
}
@CalledByNative("RTCConfiguration")
boolean getSuspendBelowMinBitrate() {
return suspendBelowMinBitrate;
}
@Nullable
@CalledByNative("RTCConfiguration")
Integer getScreencastMinBitrate() {
return screencastMinBitrate;
}
@Nullable
@CalledByNative("RTCConfiguration")
Boolean getCombinedAudioVideoBwe() {
return combinedAudioVideoBwe;
}
@Nullable
@CalledByNative("RTCConfiguration")
Boolean getEnableDtlsSrtp() {
return enableDtlsSrtp;
}
@CalledByNative("RTCConfiguration")
AdapterType getNetworkPreference() {
return networkPreference;
}
@CalledByNative("RTCConfiguration")
SdpSemantics getSdpSemantics() {
return sdpSemantics;
}
@CalledByNative("RTCConfiguration")
boolean getActiveResetSrtpParams() {
return activeResetSrtpParams;
}
@CalledByNative("RTCConfiguration")
boolean getUseMediaTransport() {
return useMediaTransport;
}
@CalledByNative("RTCConfiguration")
boolean getUseMediaTransportForDataChannels() {
return useMediaTransportForDataChannels;
}
@Nullable
@CalledByNative("RTCConfiguration")
CryptoOptions getCryptoOptions() {
return cryptoOptions;
}
@Nullable
@CalledByNative("RTCConfiguration")
String getTurnLoggingId() {
return turnLoggingId;
}
@CalledByNative("RTCConfiguration")
int getMinPort() {
return minPort;
}
@CalledByNative("RTCConfiguration")
int getMaxPort() {
return maxPort;
}
@CalledByNative("RTCConfiguration")
int getPortAllocatorFlags() {
return portAllocatorFlags;
}
};
private final List localStreams = new ArrayList<>();
private final long nativePeerConnection;
private List senders = new ArrayList<>();
private List receivers = new ArrayList<>();
private List transceivers = new ArrayList<>();
/**
* Wraps a PeerConnection created by the factory. Can be used by clients that want to implement
* their PeerConnection creation in JNI.
*/
public PeerConnection(NativePeerConnectionFactory factory) {
this(factory.createNativePeerConnection());
}
PeerConnection(long nativePeerConnection) {
this.nativePeerConnection = nativePeerConnection;
}
// JsepInterface.
public SessionDescription getLocalDescription() {
return nativeGetLocalDescription();
}
public SessionDescription getRemoteDescription() {
return nativeGetRemoteDescription();
}
public RtcCertificatePem getCertificate() {
return nativeGetCertificate();
}
public DataChannel createDataChannel(String label, DataChannel.Init init) {
return nativeCreateDataChannel(label, init);
}
public void createOffer(SdpObserver observer, MediaConstraints constraints) {
nativeCreateOffer(observer, constraints);
}
public void createAnswer(SdpObserver observer, MediaConstraints constraints) {
nativeCreateAnswer(observer, constraints);
}
public void setLocalDescription(SdpObserver observer, SessionDescription sdp) {
nativeSetLocalDescription(observer, sdp);
}
public void setRemoteDescription(SdpObserver observer, SessionDescription sdp) {
nativeSetRemoteDescription(observer, sdp);
}
/**
* 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.
*/
public void setAudioPlayout(boolean playout) {
nativeSetAudioPlayout(playout);
}
/**
* 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.
*/
public void setAudioRecording(boolean recording) {
nativeSetAudioRecording(recording);
}
public boolean setConfiguration(RTCConfiguration config) {
return nativeSetConfiguration(config);
}
public boolean addIceCandidate(IceCandidate candidate) {
return nativeAddIceCandidate(candidate.sdpMid, candidate.sdpMLineIndex, candidate.sdp);
}
public boolean removeIceCandidates(final IceCandidate[] candidates) {
return nativeRemoveIceCandidates(candidates);
}
/**
* 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.
*/
public boolean addStream(MediaStream stream) {
boolean ret = nativeAddLocalStream(stream.getNativeMediaStream());
if (!ret) {
return false;
}
localStreams.add(stream);
return true;
}
/**
* Removes the given media stream from this peer connection.
* This method is not supported with SdpSemantics.UNIFIED_PLAN. Please use
* removeTrack instead.
*/
public void removeStream(MediaStream stream) {
nativeRemoveLocalStream(stream.getNativeMediaStream());
localStreams.remove(stream);
}
/**
* 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:
*
* {@code
* 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.
*
* @param kind Corresponds to MediaStreamTrack kinds (must be "audio" or
* "video").
* @param 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.
* @return A new RtpSender object if successful, or null otherwise.
*/
public RtpSender createSender(String kind, String stream_id) {
RtpSender newSender = nativeCreateSender(kind, stream_id);
if (newSender != null) {
senders.add(newSender);
}
return newSender;
}
/**
* Gets all RtpSenders associated with this peer connection.
* Note that calling getSenders will dispose of the senders previously
* returned.
*/
public List getSenders() {
for (RtpSender sender : senders) {
sender.dispose();
}
senders = nativeGetSenders();
return Collections.unmodifiableList(senders);
}
/**
* Gets all RtpReceivers associated with this peer connection.
* Note that calling getReceivers will dispose of the receivers previously
* returned.
*/
public List getReceivers() {
for (RtpReceiver receiver : receivers) {
receiver.dispose();
}
receivers = nativeGetReceivers();
return Collections.unmodifiableList(receivers);
}
/**
* 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.
*/
public List getTransceivers() {
for (RtpTransceiver transceiver : transceivers) {
transceiver.dispose();
}
transceivers = nativeGetTransceivers();
return Collections.unmodifiableList(transceivers);
}
/**
* 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 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.
*/
public RtpSender addTrack(MediaStreamTrack track) {
return addTrack(track, Collections.emptyList());
}
public RtpSender addTrack(MediaStreamTrack track, List streamIds) {
if (track == null || streamIds == null) {
throw new NullPointerException("No MediaStreamTrack specified in addTrack.");
}
RtpSender newSender = nativeAddTrack(track.getNativeMediaStreamTrack(), streamIds);
if (newSender == null) {
throw new IllegalStateException("C++ addTrack failed.");
}
senders.add(newSender);
return newSender;
}
/**
* 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.
*/
public boolean removeTrack(RtpSender sender) {
if (sender == null) {
throw new NullPointerException("No RtpSender specified for removeTrack.");
}
return nativeRemoveTrack(sender.getNativeRtpSender());
}
/**
* 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 IllegalStateException if an error accors in C++ addTransceiver
*/
public RtpTransceiver addTransceiver(MediaStreamTrack track) {
return addTransceiver(track, new RtpTransceiver.RtpTransceiverInit());
}
public RtpTransceiver addTransceiver(
MediaStreamTrack track, @Nullable RtpTransceiver.RtpTransceiverInit init) {
if (track == null) {
throw new NullPointerException("No MediaStreamTrack specified for addTransceiver.");
}
if (init == null) {
init = new RtpTransceiver.RtpTransceiverInit();
}
RtpTransceiver newTransceiver =
nativeAddTransceiverWithTrack(track.getNativeMediaStreamTrack(), init);
if (newTransceiver == null) {
throw new IllegalStateException("C++ addTransceiver failed.");
}
transceivers.add(newTransceiver);
return newTransceiver;
}
public RtpTransceiver addTransceiver(MediaStreamTrack.MediaType mediaType) {
return addTransceiver(mediaType, new RtpTransceiver.RtpTransceiverInit());
}
public RtpTransceiver addTransceiver(
MediaStreamTrack.MediaType mediaType, @Nullable RtpTransceiver.RtpTransceiverInit init) {
if (mediaType == null) {
throw new NullPointerException("No MediaType specified for addTransceiver.");
}
if (init == null) {
init = new RtpTransceiver.RtpTransceiverInit();
}
RtpTransceiver newTransceiver = nativeAddTransceiverOfType(mediaType, init);
if (newTransceiver == null) {
throw new IllegalStateException("C++ addTransceiver failed.");
}
transceivers.add(newTransceiver);
return newTransceiver;
}
// Older, non-standard implementation of getStats.
@Deprecated
public boolean getStats(StatsObserver observer, @Nullable MediaStreamTrack track) {
return nativeOldGetStats(observer, (track == null) ? 0 : track.getNativeMediaStreamTrack());
}
/**
* 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.
*/
public void getStats(RTCStatsCollectorCallback callback) {
nativeNewGetStats(callback);
}
/**
* Limits the bandwidth allocated for all RTP streams sent by this
* PeerConnection. Pass null to leave a value unchanged.
*/
public boolean setBitrate(Integer min, Integer current, Integer max) {
return nativeSetBitrate(min, current, max);
}
/**
* 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.
*/
public boolean startRtcEventLog(int file_descriptor, int max_size_bytes) {
return nativeStartRtcEventLog(file_descriptor, max_size_bytes);
}
/**
* Stops recording an RTC event log. If no RTC event log is currently being
* recorded, this call will have no effect.
*/
public void stopRtcEventLog() {
nativeStopRtcEventLog();
}
// TODO(fischman): add support for DTMF-related methods once that API
// stabilizes.
public SignalingState signalingState() {
return nativeSignalingState();
}
public IceConnectionState iceConnectionState() {
return nativeIceConnectionState();
}
public PeerConnectionState connectionState() {
return nativeConnectionState();
}
public IceGatheringState iceGatheringState() {
return nativeIceGatheringState();
}
public void close() {
nativeClose();
}
/**
* 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.
*/
public void dispose() {
close();
for (MediaStream stream : localStreams) {
nativeRemoveLocalStream(stream.getNativeMediaStream());
stream.dispose();
}
localStreams.clear();
for (RtpSender sender : senders) {
sender.dispose();
}
senders.clear();
for (RtpReceiver receiver : receivers) {
receiver.dispose();
}
for (RtpTransceiver transceiver : transceivers) {
transceiver.dispose();
}
transceivers.clear();
receivers.clear();
nativeFreeOwnedPeerConnection(nativePeerConnection);
}
/** Returns a pointer to the native webrtc::PeerConnectionInterface. */
public long getNativePeerConnection() {
return nativeGetNativePeerConnection();
}
@CalledByNative
long getNativeOwnedPeerConnection() {
return nativePeerConnection;
}
public static long createNativePeerConnectionObserver(Observer observer) {
return nativeCreatePeerConnectionObserver(observer);
}
private native long nativeGetNativePeerConnection();
private native SessionDescription nativeGetLocalDescription();
private native SessionDescription nativeGetRemoteDescription();
private native RtcCertificatePem nativeGetCertificate();
private native DataChannel nativeCreateDataChannel(String label, DataChannel.Init init);
private native void nativeCreateOffer(SdpObserver observer, MediaConstraints constraints);
private native void nativeCreateAnswer(SdpObserver observer, MediaConstraints constraints);
private native void nativeSetLocalDescription(SdpObserver observer, SessionDescription sdp);
private native void nativeSetRemoteDescription(SdpObserver observer, SessionDescription sdp);
private native void nativeSetAudioPlayout(boolean playout);
private native void nativeSetAudioRecording(boolean recording);
private native boolean nativeSetBitrate(Integer min, Integer current, Integer max);
private native SignalingState nativeSignalingState();
private native IceConnectionState nativeIceConnectionState();
private native PeerConnectionState nativeConnectionState();
private native IceGatheringState nativeIceGatheringState();
private native void nativeClose();
private static native long nativeCreatePeerConnectionObserver(Observer observer);
private static native void nativeFreeOwnedPeerConnection(long ownedPeerConnection);
private native boolean nativeSetConfiguration(RTCConfiguration config);
private native boolean nativeAddIceCandidate(
String sdpMid, int sdpMLineIndex, String iceCandidateSdp);
private native boolean nativeRemoveIceCandidates(final IceCandidate[] candidates);
private native boolean nativeAddLocalStream(long stream);
private native void nativeRemoveLocalStream(long stream);
private native boolean nativeOldGetStats(StatsObserver observer, long nativeTrack);
private native void nativeNewGetStats(RTCStatsCollectorCallback callback);
private native RtpSender nativeCreateSender(String kind, String stream_id);
private native List nativeGetSenders();
private native List nativeGetReceivers();
private native List nativeGetTransceivers();
private native RtpSender nativeAddTrack(long track, List streamIds);
private native boolean nativeRemoveTrack(long sender);
private native RtpTransceiver nativeAddTransceiverWithTrack(
long track, RtpTransceiver.RtpTransceiverInit init);
private native RtpTransceiver nativeAddTransceiverOfType(
MediaStreamTrack.MediaType mediaType, RtpTransceiver.RtpTransceiverInit init);
private native boolean nativeStartRtcEventLog(int file_descriptor, int max_size_bytes);
private native void nativeStopRtcEventLog();
}