webrtc-wpt: use addTrack(track, stream) to increase firefox compat

Firefox does not support addTrack(track). Add a track whereever it makes sense.
The addTrack tests itself might rightfully use this.

The following grep shows most affected places:
  git grep addTrack *.html | grep -v , | grep -v "\.\.\."

Bug: None
Change-Id: Ib225e6d51184c3ccc446ccf93447e2ac7be080c3
Reviewed-on: https://chromium-review.googlesource.com/c/1346394Reviewed-by: Henrik Boström <hbos@chromium.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#611262}

third_party/blink/web_tests/external/wpt/webrtc/RTCDTMFSender-insertDTMF.https.html

@@ -122,7 +122,7 @@
     const stream = await navigator.mediaDevices.getUserMedia({audio: true});
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
-    callee.addTrack(track);
+    callee.addTrack(track, stream);
     const answer = await callee.createAnswer();
     await callee.setLocalDescription(answer);
     await caller.setRemoteDescription(answer);

third_party/blink/web_tests/external/wpt/webrtc/RTCPeerConnection-removeTrack.https.html

@@ -175,7 +175,7 @@
     const offer = await caller.createOffer();
     await caller.setLocalDescription(offer);
     await callee.setRemoteDescription(offer);
-    callee.addTrack(track);
+    callee.addTrack(track, stream);
     const answer = await callee.createAnswer();
     await callee.setLocalDescription(answer);
     await caller.setRemoteDescription(answer);
@@ -244,7 +244,7 @@
     const offer = await caller.createOffer();
     await caller.setLocalDescription(offer);
     await callee.setRemoteDescription(offer);
-    callee.addTrack(track);
+    callee.addTrack(track, stream);
     const answer = await callee.createAnswer();
     await callee.setLocalDescription(answer);
     await caller.setRemoteDescription(answer);

third_party/blink/web_tests/external/wpt/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html

@@ -228,7 +228,7 @@
     const localStream =
         await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
-    caller.addTrack(localStream.getTracks()[0]);
+    caller.addTrack(localStream.getTracks()[0], localStream);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {
       assert_array_equals(callee.getReceivers(), [e.receiver],
                           'getReceivers() == [e.receiver].');
@@ -245,7 +245,7 @@
     const localStream =
         await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
-    const sender = caller.addTrack(localStream.getTracks()[0]);
+    const sender = caller.addTrack(localStream.getTracks()[0], localStream);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track');
     await exchangeOfferAnswer(caller, callee);
     await ontrackPromise;

third_party/blink/web_tests/external/wpt/webrtc/RTCPeerConnection-track-stats.https.html

@@ -26,7 +26,7 @@
     return getUserMediaTracksAndStreams(1)
     .then(t.step_func(([tracks, streams]) => {
       track = tracks[0];
-      pc.addTrack(track);
+      pc.addTrack(track, streams[0]);
       return pc.getStats();
     }))
     .then(t.step_func(report => {
@@ -73,7 +73,7 @@
     return getUserMediaTracksAndStreams(1)
     .then(t.step_func(([tracks, streams]) => {
       track = tracks[0];
-      pc.addTrack(track);
+      pc.addTrack(track, streams[0]);
       return pc.createOffer();
     }))
     .then(t.step_func(offer => {
@@ -204,7 +204,7 @@
     return getUserMediaTracksAndStreams(1)
     .then(t.step_func(([tracks, streams]) => {
       sendingTrack = tracks[0];
-      caller.addTrack(sendingTrack);
+      caller.addTrack(sendingTrack, streams[0]);
       return doSignalingHandshake(caller, callee);
     }))
     .then(t.step_func(() => {
@@ -237,7 +237,7 @@
     };
     return getUserMediaTracksAndStreams(1)
     .then(t.step_func(([tracks, streams]) => {
-      caller.addTrack(tracks[0]);
+      caller.addTrack(tracks[0], streams[0]);
       return doSignalingHandshake(caller, callee);
     }))
     .then(t.step_func(() => {
@@ -271,7 +271,7 @@
     .then(t.step_func(([tracks, streams]) => {
       sendingTrack1 = tracks[0];
       sendingTrack2 = tracks[1];
-      sender = caller.addTrack(sendingTrack1);
+      sender = caller.addTrack(sendingTrack1, streams[0]);
       return sender.replaceTrack(sendingTrack2);
     }))
     .then(t.step_func(() => {
@@ -300,7 +300,7 @@
     .then(t.step_func(([tracks, streams]) => {
       sendingTrack1 = tracks[0];
       sendingTrack2 = tracks[1];
-      sender = caller.addTrack(sendingTrack1);
+      sender = caller.addTrack(sendingTrack1, streams[0]);
       return exchangeOffer(caller, callee);
     }))
     .then(t.step_func(() => {
@@ -337,7 +337,7 @@
     .then(t.step_func(([tracks, streams]) => {
       sendingTrack1 = tracks[0];
       sendingTrack2 = tracks[1];
-      sender = caller.addTrack(sendingTrack1);
+      sender = caller.addTrack(sendingTrack1, streams[0]);
       return doSignalingHandshake(caller, callee);
     }))
     .then(t.step_func(() => {
@@ -373,7 +373,7 @@
     .then(t.step_func(([tracks, streams]) => {
       sendingTrack1 = tracks[0];
       sendingTrack2 = tracks[1];
-      sender = caller.addTrack(sendingTrack1);
+      sender = caller.addTrack(sendingTrack1, streams[0]);
       return doSignalingHandshake(caller, callee);
     }))
     .then(t.step_func(() => {
@@ -569,8 +569,8 @@
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     let [tracks, streams] = await getUserMediaTracksAndStreams(2);
-    let sender1 = pc.addTrack(tracks[0]);
-    let sender2 = pc.addTrack(tracks[1]);
+    let sender1 = pc.addTrack(tracks[0], streams[0]);
+    let sender2 = pc.addTrack(tracks[1], streams[1]);
     await sender2.replaceTrack(sender1.track);
     await promise_rejects(t, 'InvalidAccessError', pc.getStats(sender1.track));
   }, 'RTCPeerConnection.getStats(track) throws InvalidAccessError when there ' +

third_party/blink/web_tests/external/wpt/webrtc/RTCRtpReceiver-getStats.https.html

@@ -53,7 +53,7 @@
     const stream = await getNoiseStream({audio:true});
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
-    callee.addTrack(track);
+    callee.addTrack(track, stream);
 
     const { receiver } = caller.addTransceiver('audio');