Updating addTrack test for multiple stream case and removing check.

Bug: webrtc:7932
Change-Id: I6d9a97754e6f19f5917927a24652786a4d9624c5
Reviewed-on: https://chromium-review.googlesource.com/986509
Commit-Queue: Seth Hampson <shampson@chromium.org>
Reviewed-by: Henrik Boström <hbos@chromium.org>
Cr-Commit-Position: refs/heads/master@{#548546}

third_party/WebKit/LayoutTests/fast/peerconnection/RTCPeerConnection-AddRemoveTrack.html

@@ -44,25 +44,23 @@ promise_test(function() {
     });
 }, 'addTrack() for a single track and a different stream.');
 
-promise_test(function() {
-  let pc = new RTCPeerConnection();
-  return createStreams({audio:true, video:false}, 2)
-    .then(function(streams) {
-      let streamA = streams[0];
-      let streamB = streams[1];
-      let track = streamA.getAudioTracks()[0];
-      let exception = null;
-      try {
-        pc.addTrack(track, streamA, streamB);
-      } catch (e) {
-        exception = e;
-      }
-      // The spec supports multiple streams per track but our implementation
-      // doesn't. Fix test when resolving https://crbug.com/webrtc/7932.
-      assert_true(exception != null);
-      assert_equals('NotSupportedError', exception.name);
-    });
-}, 'addTrack() for a single track and two streams throws NotSupportedError.');
+promise_test(async t => {
+  let pc = new RTCPeerConnection({sdpSemantics: "plan-b"});
+  let [streamA, streamB] = await createStreams({audio:true, video:false}, 2);
+  let track = streamA.getAudioTracks()[0];
+  assert_throws('NotSupportedError', () => {
+    pc.addTrack(track, streamA, streamB)
+  });
+}, 'addTrack() for a single track and two streams (Plan B) throws NotSupportedError.');
+
+promise_test(async t => {
+  let pc = new RTCPeerConnection({sdpSemantics: "unified-plan"});
+  let [streamA, streamB] = await createStreams({audio:true, video:false}, 2);
+  let track = streamA.getAudioTracks()[0];
+  let sender =  pc.addTrack(track, streamA, streamB);
+  assert_equals(sender.track, track);
+  assert_array_equals(pc.getLocalStreams(), [ streamA, streamB ]);
+}, 'addTrack() for a single track and two streams (Unified Plan).');
 
 promise_test(function() {
   let pc = new RTCPeerConnection();

third_party/WebKit/Source/modules/peerconnection/RTCPeerConnection.cpp

@@ -535,7 +535,8 @@ RTCPeerConnection::RTCPeerConnection(ExecutionContext* context,
       negotiation_needed_(false),
       stopped_(false),
       closed_(false),
-      has_data_channels_(false) {
+      has_data_channels_(false),
+      sdp_semantics_(configuration.sdp_semantics) {
   Document* document = ToDocument(GetExecutionContext());
 
   // If we fail, set |m_closed| and |m_stopped| to true, to avoid hitting the
@@ -1424,9 +1425,13 @@ RTCRtpSender* RTCPeerConnection::addTrack(MediaStreamTrack* track,
   DCHECK(track->Component());
   if (ThrowExceptionIfSignalingStateClosed(signaling_state_, exception_state))
     return nullptr;
-  if (streams.size() >= 2) {
-    // TODO(hbos): Don't throw an exception when this is supported by the lower
-    // layers. https://crbug.com/webrtc/7932
+  // TODO(bugs.webrtc.org/8530): Take out WebRTCSdpSemantics::kDefault check
+  // once default is no longer interpreted as Plan B lower down.
+  if ((sdp_semantics_ == WebRTCSdpSemantics::kPlanB ||
+       sdp_semantics_ == WebRTCSdpSemantics::kDefault) &&
+      streams.size() >= 2) {
+    // TODO(hbos): Update peer_handler_ to call the AddTrack() that returns the
+    // appropriate errors, and let the lower layers handle it.
     exception_state.ThrowDOMException(
         kNotSupportedError,
         "Adding a track to multiple streams is not supported.");

third_party/WebKit/Source/modules/peerconnection/RTCPeerConnection.h

@@ -44,6 +44,7 @@
 #include "platform/heap/HeapAllocator.h"
 #include "platform/scheduler/public/frame_scheduler.h"
 #include "public/platform/WebMediaConstraints.h"
+#include "public/platform/WebRTCConfiguration.h"
 #include "public/platform/WebRTCPeerConnectionHandler.h"
 #include "public/platform/WebRTCPeerConnectionHandlerClient.h"
 
@@ -351,6 +352,7 @@ class MODULES_EXPORT RTCPeerConnection final
   String last_answer_;
 
   bool has_data_channels_;  // For RAPPOR metrics
+  WebRTCSdpSemantics sdp_semantics_;
 };
 
 }  // namespace blink