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Add new PeerConnection APIs to the Java SDK.

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This adds wrappers to the following native APIs:
- SdpSemantics enum added to the RTCConfiguration
- RtpTransceiver
- PeerConnection.addTrack
- PeerConnection.removeTrack
- PeerConnection.addTransceiver
- PeerConnection.getTransceivers
These APIs are used with the new Unified Plan semantics.

Bug: webrtc:8869
Change-Id: I19443f3ff7ffc91a139ad8276331f09e57cec554
Reviewed-on: https://webrtc-review.googlesource.com/57800
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22317}
This commit is contained in:
Seth Hampson
2018-03-06 15:47:10 -08:00
committed by Commit Bot
parent 513449eab9
commit c384e14707
12 changed files with 799 additions and 63 deletions
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12
sdk/android/api/org/webrtc/MediaStream.java
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@ -88,13 +88,19 @@ public class MediaStream {
JniCommon.nativeReleaseRef(nativeStream);
}
// TODO(shampson): Remove this when downstreams have moved to using id() instead.
@Deprecated
public String label() {
return nativeGetLabel(nativeStream);
return nativeGetId(nativeStream);
}
public String getId() {
return nativeGetId(nativeStream);
}
@Override
public String toString() {
return "[" + label() + ":A=" + audioTracks.size() + ":V=" + videoTracks.size() + "]";
return "[" + getId() + ":A=" + audioTracks.size() + ":V=" + videoTracks.size() + "]";
}
@CalledByNative
@ -137,5 +143,5 @@ public class MediaStream {
long stream, long nativeVideoTrack);
private static native boolean nativeRemoveAudioTrack(long stream, long nativeAudioTrack);
private static native boolean nativeRemoveVideoTrack(long stream, long nativeVideoTrack);
private static native String nativeGetLabel(long stream);
private static native String nativeGetId(long stream);
}

270
sdk/android/api/org/webrtc/PeerConnection.java
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@ -341,6 +341,33 @@ public class PeerConnection {
}
}
/**
* 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.
*
* <p>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.
*
* <p>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.
*
* <p>For users who wish to send multiple audio/video streams and need to stay
* interoperable with legacy WebRTC implementations, specify PLAN_B.
*
* <p>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 {
@ -386,6 +413,7 @@ public class PeerConnection {
// 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.
public TurnCustomizer turnCustomizer;
@ -423,6 +451,7 @@ public class PeerConnection {
combinedAudioVideoBwe = null;
enableDtlsSrtp = null;
networkPreference = AdapterType.UNKNOWN;
sdpSemantics = SdpSemantics.PLAN_B;
}
@CalledByNative("RTCConfiguration")
@ -574,12 +603,18 @@ public class PeerConnection {
AdapterType getNetworkPreference() {
return networkPreference;
}
@CalledByNative("RTCConfiguration")
SdpSemantics getSdpSemantics() {
return sdpSemantics;
}
};
private final List<MediaStream> localStreams = new ArrayList<>();
private final long nativePeerConnection;
private List<RtpSender> senders = new ArrayList<>();
private List<RtpReceiver> receivers = new ArrayList<>();
private List<RtpTransceiver> transceivers = new ArrayList<>();
/**
* Wraps a PeerConnection created by the factory. Can be used by clients that want to implement
@ -622,18 +657,24 @@ public class PeerConnection {
nativeSetRemoteDescription(observer, sdp);
}
// True if remote audio should be played out. Defaults to true.
// 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.
/**
* 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);
}
// True if local audio shall be recorded. Defaults to true.
// 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.
/**
* 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);
}
@ -650,6 +691,11 @@ public class PeerConnection {
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.nativeStream);
if (!ret) {
@ -659,6 +705,11 @@ public class PeerConnection {
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.nativeStream);
localStreams.remove(stream);
@ -666,15 +717,15 @@ public class PeerConnection {
/**
* Creates an RtpSender without a track.
* <p>
* This method allows an application to cause the PeerConnection to negotiate
*
* <p>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.
* <p>
* When the application does want to begin sending a track, it can call
*
* <p>When the application does want to begin sending a track, it can call
* RtpSender.setTrack, which doesn't require any additional SDP negotiation.
* <p>
* Example use:
*
* <p>Example use:
* <pre>
* {@code
* audioSender = pc.createSender("audio", "stream1");
@ -686,12 +737,15 @@ public class PeerConnection {
* videoSender.setTrack(videoTrack, false);
* }
* </pre>
* Note: This corresponds most closely to "addTransceiver" in the official
* <p>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.
*
* <p>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
@ -702,15 +756,18 @@ public class PeerConnection {
* @return A new RtpSender object if successful, or null otherwise.
*/
public RtpSender createSender(String kind, String stream_id) {
RtpSender new_sender = nativeCreateSender(kind, stream_id);
if (new_sender != null) {
senders.add(new_sender);
RtpSender newSender = nativeCreateSender(kind, stream_id);
if (newSender != null) {
senders.add(newSender);
}
return new_sender;
return newSender;
}
// Note that calling getSenders will dispose of the senders previously
// returned (and same goes for getReceivers).
/**
* Gets all RtpSenders associated with this peer connection.
* Note that calling getSenders will dispose of the senders previously
* returned.
*/
public List<RtpSender> getSenders() {
for (RtpSender sender : senders) {
sender.dispose();
@ -719,6 +776,11 @@ public class PeerConnection {
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<RtpReceiver> getReceivers() {
for (RtpReceiver receiver : receivers) {
receiver.dispose();
@ -727,35 +789,162 @@ public class PeerConnection {
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<RtpTransceiver> 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<String> streamIds) {
if (track == null || streamIds == null) {
throw new NullPointerException("No MediaStreamTrack specified in addTrack.");
}
RtpSender newSender = nativeAddTrack(track.nativeTrack, 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.nativeRtpSender);
}
/**
* 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.
*
* <p>The initial value of |mid| in the returned transceiver is null. Setting a
* new session description may change it to a non-null value.
*
* <p>https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
*
* <p>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.
*
* <p>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.
*
* <p>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.
*
* <p>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, 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.nativeTrack, 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, 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, MediaStreamTrack track) {
return nativeOldGetStats(observer, (track == null) ? 0 : track.nativeTrack);
}
// 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.
/**
* 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.
/**
* 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.
/**
* 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.
/**
* 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();
}
@ -780,14 +969,14 @@ public class PeerConnection {
/**
* Free native resources associated with this PeerConnection instance.
* <p>
*
* 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.
* <p>
* Note that this method cannot be safely called from an observer callback
*
* <p>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
@ -808,6 +997,10 @@ public class PeerConnection {
for (RtpReceiver receiver : receivers) {
receiver.dispose();
}
for (RtpTransceiver transceiver : transceivers) {
transceiver.dispose();
}
transceivers.clear();
receivers.clear();
nativeFreeOwnedPeerConnection(nativePeerConnection);
}
@ -854,6 +1047,13 @@ public class PeerConnection {
private native RtpSender nativeCreateSender(String kind, String stream_id);
private native List<RtpSender> nativeGetSenders();
private native List<RtpReceiver> nativeGetReceivers();
private native List<RtpTransceiver> nativeGetTransceivers();
private native RtpSender nativeAddTrack(long track, List<String> 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();
}

215
sdk/android/api/org/webrtc/RtpTransceiver.java Normal file
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@ -0,0 +1,215 @@
/*
* Copyright 2018 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.Collections;
import java.util.List;
import org.webrtc.RtpParameters.Encoding;
/**
* Java wrapper for a C++ RtpTransceiverInterface.
*
* <p>The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the WebRTC
* specification. A transceiver represents a combination of an RTCRtpSender
* and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
* RTCRtpTransceiver is said to be associated with a media description if its
* mid property is non-nil; otherwise, it is said to be disassociated.
* JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
*
* <p>Note that RTCRtpTransceivers are only supported when using
* RTCPeerConnection with Unified Plan SDP.
*
* <p>WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
*/
@JNINamespace("webrtc::jni")
public class RtpTransceiver {
/** Java version of webrtc::RtpTransceiverDirection - the ordering must be kept in sync. */
public enum RtpTransceiverDirection {
SEND_RECV(0),
SEND_ONLY(1),
RECV_ONLY(2),
INACTIVE(3);
private final int nativeIndex;
private RtpTransceiverDirection(int nativeIndex) {
this.nativeIndex = nativeIndex;
}
@CalledByNative("RtpTransceiverDirection")
int getNativeIndex() {
return nativeIndex;
}
@CalledByNative("RtpTransceiverDirection")
static RtpTransceiverDirection fromNativeIndex(int nativeIndex) {
for (RtpTransceiverDirection type : RtpTransceiverDirection.values()) {
if (type.getNativeIndex() == nativeIndex) {
return type;
}
}
throw new IllegalArgumentException(
"Uknown native RtpTransceiverDirection type" + nativeIndex);
}
}
/**
* Tracks webrtc::RtpTransceiverInit. https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
* A structure for initializing an RtpTransceiver in a call to addTransceiver.
* Note: This does not contain a list of encoding parameters, because they are currently
* not being used natively.
*/
public static final class RtpTransceiverInit {
private final RtpTransceiverDirection direction;
private final List<String> streamIds;
public RtpTransceiverInit() {
this(RtpTransceiverDirection.SEND_RECV);
}
public RtpTransceiverInit(RtpTransceiverDirection direction) {
this(direction, Collections.emptyList());
}
public RtpTransceiverInit(RtpTransceiverDirection direction, List<String> streamIds) {
this.direction = direction;
this.streamIds = new ArrayList<String>(streamIds);
}
@CalledByNative("RtpTransceiverInit")
int getDirectionNativeIndex() {
return direction.getNativeIndex();
}
@CalledByNative("RtpTransceiverInit")
List<String> getStreamIds() {
return new ArrayList<String>(this.streamIds);
}
}
private final long nativeRtpTransceiver;
private RtpSender cachedSender;
private RtpReceiver cachedReceiver;
@CalledByNative
protected RtpTransceiver(long nativeRtpTransceiver) {
this.nativeRtpTransceiver = nativeRtpTransceiver;
cachedSender = nativeGetSender(nativeRtpTransceiver);
cachedReceiver = nativeGetReceiver(nativeRtpTransceiver);
}
/**
* Media type of the transceiver. Any sender(s)/receiver(s) will have this
* type as well.
*/
public MediaStreamTrack.MediaType getMediaType() {
return nativeGetMediaType(nativeRtpTransceiver);
}
/**
* The mid attribute is the mid negotiated and present in the local and
* remote descriptions. Before negotiation is complete, the mid value may be
* null. After rollbacks, the value may change from a non-null value to null.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
*/
public String getMid() {
return nativeGetMid(nativeRtpTransceiver);
}
/**
* The sender attribute exposes the RtpSender corresponding to the RTP media
* that may be sent with the transceiver's mid. The sender is always present,
* regardless of the direction of media.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
*/
public RtpSender getSender() {
return cachedSender;
}
/**
* The receiver attribute exposes the RtpReceiver corresponding to the RTP
* media that may be received with the transceiver's mid. The receiver is
* always present, regardless of the direction of media.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
*/
public RtpReceiver getReceiver() {
return cachedReceiver;
}
/**
* The stopped attribute indicates that the sender of this transceiver will no
* longer send, and that the receiver will no longer receive. It is true if
* either stop has been called or if setting the local or remote description
* has caused the RtpTransceiver to be stopped.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
*/
public boolean isStopped() {
return nativeStopped(nativeRtpTransceiver);
}
/**
* The direction attribute indicates the preferred direction of this
* transceiver, which will be used in calls to CreateOffer and CreateAnswer.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
*/
public RtpTransceiverDirection getDirection() {
return nativeDirection(nativeRtpTransceiver);
}
/**
* The current_direction attribute indicates the current direction negotiated
* for this transceiver. If this transceiver has never been represented in an
* offer/answer exchange, or if the transceiver is stopped, the value is null.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
*/
public RtpTransceiverDirection getCurrentDirection() {
return nativeCurrentDirection(nativeRtpTransceiver);
}
/**
* Sets the preferred direction of this transceiver. An update of
* directionality does not take effect immediately. Instead, future calls to
* CreateOffer and CreateAnswer mark the corresponding media descriptions as
* sendrecv, sendonly, recvonly, or inactive.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
*/
public void setDirection(RtpTransceiverDirection rtpTransceiverDirection) {
nativeSetDirection(nativeRtpTransceiver, rtpTransceiverDirection);
}
/**
* The Stop method irreversibly stops the RtpTransceiver. The sender of this
* transceiver will no longer send, the receiver will no longer receive.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
*/
public void stop() {
nativeStop(nativeRtpTransceiver);
}
public void dispose() {
cachedSender.dispose();
cachedReceiver.dispose();
JniCommon.nativeReleaseRef(nativeRtpTransceiver);
}
private static native MediaStreamTrack.MediaType nativeGetMediaType(long rtpTransceiver);
private static native String nativeGetMid(long rtpTransceiver);
private static native RtpSender nativeGetSender(long rtpTransceiver);
private static native RtpReceiver nativeGetReceiver(long rtpTransceiver);
private static native boolean nativeStopped(long rtpTransceiver);
private static native RtpTransceiverDirection nativeDirection(long rtpTransceiver);
private static native RtpTransceiverDirection nativeCurrentDirection(long rtpTransceiver);
private static native void nativeStop(long rtpTransceiver);
private static native void nativeSetDirection(
long rtpTransceiver, RtpTransceiverDirection rtpTransceiverDirection);
}