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Java binding for RtpSender/RtpReceiver.

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The Java PeerConnection maintains a cached list of Java RtpSenders
and RtpReceivers so that the same objects are returned every time
getSenders() or getReceivers() is called. They are disposed of when
PeerConnection.dispose() is called, which will also dispose their
referenced MediaStreamTracks.

Review URL: https://codereview.webrtc.org/1368143002

Cr-Commit-Position: refs/heads/master@{#10189}
This commit is contained in:
deadbeef
2015-10-06 12:29:25 -07:00
committed by Commit bot
parent 10950692d6
commit 4139c0f1c5
7 changed files with 295 additions and 1 deletions
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3
talk/app/webrtc/java/jni/classreferenceholder.cc
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@ -63,6 +63,7 @@ void FreeGlobalClassReferenceHolder() {
ClassReferenceHolder::ClassReferenceHolder(JNIEnv* jni) {
LoadClass(jni, "java/nio/ByteBuffer");
LoadClass(jni, "java/util/ArrayList");
LoadClass(jni, "org/webrtc/AudioTrack");
LoadClass(jni, "org/webrtc/DataChannel");
LoadClass(jni, "org/webrtc/DataChannel$Buffer");
@ -107,6 +108,8 @@ ClassReferenceHolder::ClassReferenceHolder(JNIEnv* jni) {
LoadClass(jni, "org/webrtc/PeerConnection$TcpCandidatePolicy");
LoadClass(jni, "org/webrtc/PeerConnection$KeyType");
LoadClass(jni, "org/webrtc/PeerConnection$SignalingState");
LoadClass(jni, "org/webrtc/RtpReceiver");
LoadClass(jni, "org/webrtc/RtpSender");
LoadClass(jni, "org/webrtc/SessionDescription");
LoadClass(jni, "org/webrtc/SessionDescription$Type");
LoadClass(jni, "org/webrtc/StatsReport");

109
talk/app/webrtc/java/jni/peerconnection_jni.cc
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@ -64,6 +64,8 @@
#include "talk/app/webrtc/dtlsidentitystore.h"
#include "talk/app/webrtc/mediaconstraintsinterface.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/rtpreceiverinterface.h"
#include "talk/app/webrtc/rtpsenderinterface.h"
#include "talk/app/webrtc/videosourceinterface.h"
#include "talk/media/base/videocapturer.h"
#include "talk/media/base/videorenderer.h"
@ -113,6 +115,8 @@ using webrtc::MediaStreamTrackInterface;
using webrtc::PeerConnectionFactoryInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
using webrtc::RtpReceiverInterface;
using webrtc::RtpSenderInterface;
using webrtc::SessionDescriptionInterface;
using webrtc::SetSessionDescriptionObserver;
using webrtc::StatsObserver;
@ -971,7 +975,7 @@ JOW(void, VideoRenderer_releaseNativeFrame)(
}
JOW(void, MediaStreamTrack_free)(JNIEnv*, jclass, jlong j_p) {
CHECK_RELEASE(reinterpret_cast<MediaStreamTrackInterface*>(j_p));
reinterpret_cast<MediaStreamTrackInterface*>(j_p)->Release();
}
JOW(jboolean, MediaStream_nativeAddAudioTrack)(
@ -1626,6 +1630,60 @@ JOW(void, PeerConnection_nativeRemoveLocalStream)(
reinterpret_cast<MediaStreamInterface*>(native_stream));
}
JOW(jobject, PeerConnection_nativeGetSenders)(JNIEnv* jni, jobject j_pc) {
jclass j_array_list_class = FindClass(jni, "java/util/ArrayList");
jmethodID j_array_list_ctor =
GetMethodID(jni, j_array_list_class, "<init>", "()V");
jmethodID j_array_list_add =
GetMethodID(jni, j_array_list_class, "add", "(Ljava/lang/Object;)Z");
jobject j_senders = jni->NewObject(j_array_list_class, j_array_list_ctor);
CHECK_EXCEPTION(jni) << "error during NewObject";
jclass j_rtp_sender_class = FindClass(jni, "org/webrtc/RtpSender");
jmethodID j_rtp_sender_ctor =
GetMethodID(jni, j_rtp_sender_class, "<init>", "(J)V");
auto senders = ExtractNativePC(jni, j_pc)->GetSenders();
for (const auto& sender : senders) {
jlong nativeSenderPtr = jlongFromPointer(sender.get());
jobject j_sender =
jni->NewObject(j_rtp_sender_class, j_rtp_sender_ctor, nativeSenderPtr);
CHECK_EXCEPTION(jni) << "error during NewObject";
// Sender is now owned by Java object, and will be freed from there.
sender->AddRef();
jni->CallBooleanMethod(j_senders, j_array_list_add, j_sender);
CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
}
return j_senders;
}
JOW(jobject, PeerConnection_nativeGetReceivers)(JNIEnv* jni, jobject j_pc) {
jclass j_array_list_class = FindClass(jni, "java/util/ArrayList");
jmethodID j_array_list_ctor =
GetMethodID(jni, j_array_list_class, "<init>", "()V");
jmethodID j_array_list_add =
GetMethodID(jni, j_array_list_class, "add", "(Ljava/lang/Object;)Z");
jobject j_receivers = jni->NewObject(j_array_list_class, j_array_list_ctor);
CHECK_EXCEPTION(jni) << "error during NewObject";
jclass j_rtp_receiver_class = FindClass(jni, "org/webrtc/RtpReceiver");
jmethodID j_rtp_receiver_ctor =
GetMethodID(jni, j_rtp_receiver_class, "<init>", "(J)V");
auto receivers = ExtractNativePC(jni, j_pc)->GetReceivers();
for (const auto& receiver : receivers) {
jlong nativeReceiverPtr = jlongFromPointer(receiver.get());
jobject j_receiver = jni->NewObject(j_rtp_receiver_class,
j_rtp_receiver_ctor, nativeReceiverPtr);
CHECK_EXCEPTION(jni) << "error during NewObject";
// Receiver is now owned by Java object, and will be freed from there.
receiver->AddRef();
jni->CallBooleanMethod(j_receivers, j_array_list_add, j_receiver);
CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
}
return j_receivers;
}
JOW(bool, PeerConnection_nativeGetStats)(
JNIEnv* jni, jobject j_pc, jobject j_observer, jlong native_track) {
rtc::scoped_refptr<StatsObserverWrapper> observer(
@ -1871,4 +1929,53 @@ JOW(jbyteArray, CallSessionFileRotatingLogSink_nativeGetLogData)(
return result;
}
JOW(void, RtpSender_nativeSetTrack)(JNIEnv* jni,
jclass,
jlong j_rtp_sender_pointer,
jlong j_track_pointer) {
reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
->SetTrack(reinterpret_cast<MediaStreamTrackInterface*>(j_track_pointer));
}
JOW(jlong, RtpSender_nativeGetTrack)(JNIEnv* jni,
jclass,
jlong j_rtp_sender_pointer,
jlong j_track_pointer) {
return jlongFromPointer(
reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
->track()
.release());
}
JOW(jstring, RtpSender_nativeId)(
JNIEnv* jni, jclass, jlong j_rtp_sender_pointer) {
return JavaStringFromStdString(
jni, reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)->id());
}
JOW(void, RtpSender_free)(JNIEnv* jni, jclass, jlong j_rtp_sender_pointer) {
reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)->Release();
}
JOW(jlong, RtpReceiver_nativeGetTrack)(JNIEnv* jni,
jclass,
jlong j_rtp_receiver_pointer,
jlong j_track_pointer) {
return jlongFromPointer(
reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
->track()
.release());
}
JOW(jstring, RtpReceiver_nativeId)(
JNIEnv* jni, jclass, jlong j_rtp_receiver_pointer) {
return JavaStringFromStdString(
jni,
reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)->id());
}
JOW(void, RtpReceiver_free)(JNIEnv* jni, jclass, jlong j_rtp_receiver_pointer) {
reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)->Release();
}
} // namespace webrtc_jni

35
talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
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@ -29,6 +29,7 @@
package org.webrtc;
import java.util.ArrayList;
import java.util.Collections;
import java.util.LinkedList;
import java.util.List;
@ -170,11 +171,15 @@ public class PeerConnection {
private final List<MediaStream> localStreams;
private final long nativePeerConnection;
private final long nativeObserver;
private List<RtpSender> senders;
private List<RtpReceiver> receivers;
PeerConnection(long nativePeerConnection, long nativeObserver) {
this.nativePeerConnection = nativePeerConnection;
this.nativeObserver = nativeObserver;
localStreams = new LinkedList<MediaStream>();
senders = new LinkedList<RtpSender>();
receivers = new LinkedList<RtpReceiver>();
}
// JsepInterface.
@ -218,6 +223,24 @@ public class PeerConnection {
localStreams.remove(stream);
}
// Note that calling getSenders will dispose of the senders previously
// returned (and same goes for getReceivers).
public List<RtpSender> getSenders() {
for (RtpSender sender : senders) {
sender.dispose();
}
senders = nativeGetSenders();
return Collections.unmodifiableList(senders);
}
public List<RtpReceiver> getReceivers() {
for (RtpReceiver receiver : receivers) {
receiver.dispose();
}
receivers = nativeGetReceivers();
return Collections.unmodifiableList(receivers);
}
public boolean getStats(StatsObserver observer, MediaStreamTrack track) {
return nativeGetStats(observer, (track == null) ? 0 : track.nativeTrack);
}
@ -239,6 +262,14 @@ public class PeerConnection {
stream.dispose();
}
localStreams.clear();
for (RtpSender sender : senders) {
sender.dispose();
}
senders.clear();
for (RtpReceiver receiver : receivers) {
receiver.dispose();
}
receivers.clear();
freePeerConnection(nativePeerConnection);
freeObserver(nativeObserver);
}
@ -256,4 +287,8 @@ public class PeerConnection {
private native boolean nativeGetStats(
StatsObserver observer, long nativeTrack);
private native List<RtpSender> nativeGetSenders();
private native List<RtpReceiver> nativeGetReceivers();
}

63
talk/app/webrtc/java/src/org/webrtc/RtpReceiver.java Normal file
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@ -0,0 +1,63 @@
/*
* libjingle
* Copyright 2015 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
package org.webrtc;
/** Java wrapper for a C++ RtpReceiverInterface. */
public class RtpReceiver {
final long nativeRtpReceiver;
private MediaStreamTrack cachedTrack;
public RtpReceiver(long nativeRtpReceiver) {
this.nativeRtpReceiver = nativeRtpReceiver;
long track = nativeGetTrack(nativeRtpReceiver);
// We can assume that an RtpReceiver always has an associated track.
cachedTrack = new MediaStreamTrack(track);
}
public MediaStreamTrack track() {
return cachedTrack;
}
public String id() {
return nativeId(nativeRtpReceiver);
}
public void dispose() {
cachedTrack.dispose();
free(nativeRtpReceiver);
}
// This should increment the reference count of the track.
// Will be released in dispose().
private static native long nativeGetTrack(long nativeRtpReceiver);
private static native String nativeId(long nativeRtpReceiver);
private static native void free(long nativeRtpReceiver);
};

79
talk/app/webrtc/java/src/org/webrtc/RtpSender.java Normal file
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@ -0,0 +1,79 @@
/*
* libjingle
* Copyright 2015 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
package org.webrtc;
/** Java wrapper for a C++ RtpSenderInterface. */
public class RtpSender {
final long nativeRtpSender;
private MediaStreamTrack cachedTrack;
public RtpSender(long nativeRtpSender) {
this.nativeRtpSender = nativeRtpSender;
long track = nativeGetTrack(nativeRtpSender);
// It may be possible for an RtpSender to be created without a track.
cachedTrack = (track == 0) ? null : new MediaStreamTrack(track);
}
// NOTE: This should not be called with a track that's already used by
// another RtpSender, because then it would be double-disposed.
public void setTrack(MediaStreamTrack track) {
if (cachedTrack != null) {
cachedTrack.dispose();
}
cachedTrack = track;
nativeSetTrack(nativeRtpSender, (track == null) ? 0 : track.nativeTrack);
}
public MediaStreamTrack track() {
return cachedTrack;
}
public String id() {
return nativeId(nativeRtpSender);
}
public void dispose() {
if (cachedTrack != null) {
cachedTrack.dispose();
}
free(nativeRtpSender);
}
private static native void nativeSetTrack(long nativeRtpSender,
long nativeTrack);
// This should increment the reference count of the track.
// Will be released in dispose() or setTrack().
private static native long nativeGetTrack(long nativeRtpSender);
private static native String nativeId(long nativeRtpSender);
private static native void free(long nativeRtpSender);
}
;

5
talk/app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
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@ -645,6 +645,11 @@ public class PeerConnectionTest {
assertEquals(answeringPC.getLocalDescription().type, answerSdp.type);
assertEquals(answeringPC.getRemoteDescription().type, offerSdp.type);
assertEquals(offeringPC.getSenders().size(), 2);
assertEquals(offeringPC.getReceivers().size(), 2);
assertEquals(answeringPC.getSenders().size(), 2);
assertEquals(answeringPC.getReceivers().size(), 2);
if (!RENDER_TO_GUI) {
// Wait for at least some frames to be delivered at each end (number
// chosen arbitrarily).

2
talk/libjingle.gyp
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@ -134,6 +134,8 @@
'app/webrtc/java/src/org/webrtc/MediaStreamTrack.java',
'app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java',
'app/webrtc/java/src/org/webrtc/PeerConnection.java',
'app/webrtc/java/src/org/webrtc/RtpReceiver.java',
'app/webrtc/java/src/org/webrtc/RtpSender.java',
'app/webrtc/java/src/org/webrtc/SdpObserver.java',
'app/webrtc/java/src/org/webrtc/StatsObserver.java',
'app/webrtc/java/src/org/webrtc/StatsReport.java',