Fix for scenario where m-line is revived after being set to port 0.
When this is detected, we'll now "reconfigure" the senders and
receivers, which will reconnect the capturers/renderers to the
underlying streams which have been recreated.
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1428243005
Cr-Commit-Position: refs/heads/master@{#10628}
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index 3cf66d6..3193ffd 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -78,11 +78,13 @@
using webrtc::DtmfSenderObserverInterface;
using webrtc::FakeConstraints;
using webrtc::MediaConstraintsInterface;
+using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
using webrtc::MockCreateSessionDescriptionObserver;
using webrtc::MockDataChannelObserver;
using webrtc::MockSetSessionDescriptionObserver;
using webrtc::MockStatsObserver;
+using webrtc::ObserverInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionFactory;
using webrtc::SessionDescriptionInterface;
@@ -139,7 +141,8 @@
};
class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
- public SignalingMessageReceiver {
+ public SignalingMessageReceiver,
+ public ObserverInterface {
public:
static PeerConnectionTestClient* CreateClient(
const std::string& id,
@@ -206,7 +209,8 @@
webrtc::PeerConnectionInterface::SignalingState new_state) override {
EXPECT_EQ(pc()->signaling_state(), new_state);
}
- void OnAddStream(webrtc::MediaStreamInterface* media_stream) override {
+ void OnAddStream(MediaStreamInterface* media_stream) override {
+ media_stream->RegisterObserver(this);
for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) {
const std::string id = media_stream->GetVideoTracks()[i]->id();
ASSERT_TRUE(fake_video_renderers_.find(id) ==
@@ -215,7 +219,7 @@
new webrtc::FakeVideoTrackRenderer(media_stream->GetVideoTracks()[i]);
}
}
- void OnRemoveStream(webrtc::MediaStreamInterface* media_stream) override {}
+ void OnRemoveStream(MediaStreamInterface* media_stream) override {}
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
@@ -238,6 +242,40 @@
candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
}
+ // MediaStreamInterface callback
+ void OnChanged() override {
+ // Track added or removed from MediaStream, so update our renderers.
+ rtc::scoped_refptr<StreamCollectionInterface> remote_streams =
+ pc()->remote_streams();
+ // Remove renderers for tracks that were removed.
+ for (auto it = fake_video_renderers_.begin();
+ it != fake_video_renderers_.end();) {
+ if (remote_streams->FindVideoTrack(it->first) == nullptr) {
+ auto to_delete = it++;
+ delete to_delete->second;
+ fake_video_renderers_.erase(to_delete);
+ } else {
+ ++it;
+ }
+ }
+ // Create renderers for new video tracks.
+ for (size_t stream_index = 0; stream_index < remote_streams->count();
+ ++stream_index) {
+ MediaStreamInterface* remote_stream = remote_streams->at(stream_index);
+ for (size_t track_index = 0;
+ track_index < remote_stream->GetVideoTracks().size();
+ ++track_index) {
+ const std::string id =
+ remote_stream->GetVideoTracks()[track_index]->id();
+ if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) {
+ continue;
+ }
+ fake_video_renderers_[id] = new webrtc::FakeVideoTrackRenderer(
+ remote_stream->GetVideoTracks()[track_index]);
+ }
+ }
+ }
+
void SetVideoConstraints(const webrtc::FakeConstraints& video_constraint) {
video_constraints_ = video_constraint;
}
@@ -246,7 +284,7 @@
std::string stream_label =
kStreamLabelBase +
rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count()));
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+ rtc::scoped_refptr<MediaStreamInterface> stream =
peer_connection_factory_->CreateLocalMediaStream(stream_label);
if (audio && can_receive_audio()) {
@@ -276,6 +314,12 @@
return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
}
+ // Automatically add a stream when receiving an offer, if we don't have one.
+ // Defaults to true.
+ void set_auto_add_stream(bool auto_add_stream) {
+ auto_add_stream_ = auto_add_stream;
+ }
+
void set_signaling_message_receiver(
SignalingMessageReceiver* signaling_message_receiver) {
signaling_message_receiver_ = signaling_message_receiver;
@@ -705,7 +749,7 @@
void HandleIncomingOffer(const std::string& msg) {
LOG(INFO) << id_ << "HandleIncomingOffer ";
- if (NumberOfLocalMediaStreams() == 0) {
+ if (NumberOfLocalMediaStreams() == 0 && auto_add_stream_) {
// If we are not sending any streams ourselves it is time to add some.
AddMediaStream(true, true);
}
@@ -812,6 +856,8 @@
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
+ bool auto_add_stream_ = true;
+
typedef std::pair<std::string, std::string> IceUfragPwdPair;
std::map<int, IceUfragPwdPair> ice_ufrag_pwd_;
bool expect_ice_restart_ = false;
@@ -963,7 +1009,6 @@
ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
VerifySessionDescriptions();
-
int audio_frame_count = kEndAudioFrameCount;
// TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
if (!initiating_client_->can_receive_audio() ||
@@ -1562,6 +1607,29 @@
EXPECT_NE(receiver_candidate, receiver_candidate_restart);
}
+// This test sets up a call between two parties with audio, and video.
+// It then renegotiates setting the video m-line to "port 0", then later
+// renegotiates again, enabling video.
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestVideoDisableEnable) {
+ ASSERT_TRUE(CreateTestClients());
+
+ // Do initial negotiation. Will result in video and audio sendonly m-lines.
+ receiving_client()->set_auto_add_stream(false);
+ initializing_client()->AddMediaStream(true, true);
+ initializing_client()->Negotiate();
+
+ // Negotiate again, disabling the video m-line (receiving client will
+ // set port to 0 due to mandatory "OfferToReceiveVideo: false" constraint).
+ receiving_client()->SetReceiveVideo(false);
+ initializing_client()->Negotiate();
+
+ // Enable video and do negotiation again, making sure video is received
+ // end-to-end.
+ receiving_client()->SetReceiveVideo(true);
+ receiving_client()->AddMediaStream(true, true);
+ LocalP2PTest();
+}
+
// This test sets up a Jsep call between two parties with external
// VideoDecoderFactory.
// TODO(holmer): Disabled due to sometimes crashing on buildbots.
