Adding a MediaStream parameter to createSender.
This will allow an app to create senders with the same stream id,
without SDP munging.
Review URL: https://codereview.webrtc.org/1538673002
Cr-Commit-Position: refs/heads/master@{#11092}
diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/talk/app/webrtc/java/jni/peerconnection_jni.cc
index b16f945..c31859d 100644
--- a/talk/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc
@@ -1792,14 +1792,15 @@
}
JOW(jobject, PeerConnection_nativeCreateSender)(
- JNIEnv* jni, jobject j_pc, jstring j_kind) {
+ JNIEnv* jni, jobject j_pc, jstring j_kind, jstring j_stream_id) {
jclass j_rtp_sender_class = FindClass(jni, "org/webrtc/RtpSender");
jmethodID j_rtp_sender_ctor =
GetMethodID(jni, j_rtp_sender_class, "<init>", "(J)V");
std::string kind = JavaToStdString(jni, j_kind);
+ std::string stream_id = JavaToStdString(jni, j_stream_id);
rtc::scoped_refptr<RtpSenderInterface> sender =
- ExtractNativePC(jni, j_pc)->CreateSender(kind);
+ ExtractNativePC(jni, j_pc)->CreateSender(kind, stream_id);
if (!sender.get()) {
return nullptr;
}
diff --git a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java b/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
index 5f037a1..36cd075 100644
--- a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
+++ b/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
@@ -224,8 +224,8 @@
localStreams.remove(stream);
}
- public RtpSender createSender(String kind) {
- RtpSender new_sender = nativeCreateSender(kind);
+ public RtpSender createSender(String kind, String stream_id) {
+ RtpSender new_sender = nativeCreateSender(kind, stream_id);
if (new_sender != null) {
senders.add(new_sender);
}
@@ -297,7 +297,7 @@
private native boolean nativeGetStats(
StatsObserver observer, long nativeTrack);
- private native RtpSender nativeCreateSender(String kind);
+ private native RtpSender nativeCreateSender(String kind, String stream_id);
private native List<RtpSender> nativeGetSenders();
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 85e03f9..617eb15 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -813,7 +813,8 @@
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
- const std::string& kind) {
+ const std::string& kind,
+ const std::string& stream_id) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
RtpSenderInterface* new_sender;
if (kind == MediaStreamTrackInterface::kAudioKind) {
@@ -824,6 +825,9 @@
LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
return rtc::scoped_refptr<RtpSenderInterface>();
}
+ if (!stream_id.empty()) {
+ new_sender->set_stream_id(stream_id);
+ }
senders_.push_back(new_sender);
return RtpSenderProxy::Create(signaling_thread(), new_sender);
}
diff --git a/talk/app/webrtc/peerconnection.h b/talk/app/webrtc/peerconnection.h
index f6c4fc3..ab3fdcc 100644
--- a/talk/app/webrtc/peerconnection.h
+++ b/talk/app/webrtc/peerconnection.h
@@ -103,7 +103,8 @@
AudioTrackInterface* track) override;
rtc::scoped_refptr<RtpSenderInterface> CreateSender(
- const std::string& kind) override;
+ const std::string& kind,
+ const std::string& stream_id) override;
std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const override;
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index a127869..44c7e7f 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -1817,8 +1817,10 @@
// received end-to-end.
TEST_F(P2PTestConductor, EarlyWarmupTest) {
ASSERT_TRUE(CreateTestClients());
- auto audio_sender = initializing_client()->pc()->CreateSender("audio");
- auto video_sender = initializing_client()->pc()->CreateSender("video");
+ auto audio_sender =
+ initializing_client()->pc()->CreateSender("audio", "stream_id");
+ auto video_sender =
+ initializing_client()->pc()->CreateSender("video", "stream_id");
initializing_client()->Negotiate();
// Wait for ICE connection to complete, without any tracks.
// Note that the receiving client WILL (in HandleIncomingOffer) create
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index 799ca15..4648176 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -338,8 +338,11 @@
// TODO(deadbeef): Make these pure virtual once all subclasses implement them.
// |kind| must be "audio" or "video".
+ // |stream_id| is used to populate the msid attribute; if empty, one will
+ // be generated automatically.
virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
- const std::string& kind) {
+ const std::string& kind,
+ const std::string& stream_id) {
return rtc::scoped_refptr<RtpSenderInterface>();
}
diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index 930f538..098f8ae 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -1198,6 +1198,22 @@
EXPECT_TRUE(video_desc == nullptr);
}
+// Test creating a sender with a stream ID, and ensure the ID is populated
+// in the offer.
+TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
+ CreatePeerConnection();
+ pc_->CreateSender("video", kStreamLabel1);
+
+ scoped_ptr<SessionDescriptionInterface> offer;
+ ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
+
+ const cricket::MediaContentDescription* video_desc =
+ cricket::GetFirstVideoContentDescription(offer->description());
+ ASSERT_TRUE(video_desc != nullptr);
+ ASSERT_EQ(1u, video_desc->streams().size());
+ EXPECT_EQ(kStreamLabel1, video_desc->streams()[0].sync_label);
+}
+
// Test that we can specify a certain track that we want statistics about.
TEST_F(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
InitiateCall();
diff --git a/talk/app/webrtc/peerconnectionproxy.h b/talk/app/webrtc/peerconnectionproxy.h
index 9b446c0..3c983d7 100644
--- a/talk/app/webrtc/peerconnectionproxy.h
+++ b/talk/app/webrtc/peerconnectionproxy.h
@@ -43,8 +43,9 @@
PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
CreateDtmfSender, AudioTrackInterface*)
- PROXY_METHOD1(rtc::scoped_refptr<RtpSenderInterface>,
+ PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
CreateSender,
+ const std::string&,
const std::string&)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
GetSenders)
