From 8c251824b9e5949efd1fb52155b0daa41a13cb46 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Thu, 22 Nov 2018 05:00:04 -0800 Subject: [PATCH] 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 --- webrtc/RTCDTMFSender-insertDTMF.https.html | 2 +- .../RTCPeerConnection-removeTrack.https.html | 4 ++-- ...ion-setRemoteDescription-tracks.https.html | 4 ++-- .../RTCPeerConnection-track-stats.https.html | 20 +++++++++---------- webrtc/RTCRtpReceiver-getStats.https.html | 2 +- 5 files changed, 16 insertions(+), 16 deletions(-) diff --git a/webrtc/RTCDTMFSender-insertDTMF.https.html b/webrtc/RTCDTMFSender-insertDTMF.https.html index 8a6d6456f3a9e0..8ac144bab9286f 100644 --- a/webrtc/RTCDTMFSender-insertDTMF.https.html +++ b/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); diff --git a/webrtc/RTCPeerConnection-removeTrack.https.html b/webrtc/RTCPeerConnection-removeTrack.https.html index c98dbdbd8f3fa0..a955db6895ad17 100644 --- a/webrtc/RTCPeerConnection-removeTrack.https.html +++ b/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); diff --git a/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html b/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html index c9d2d1282c7b21..aa3f93e235e340 100644 --- a/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html +++ b/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; diff --git a/webrtc/RTCPeerConnection-track-stats.https.html b/webrtc/RTCPeerConnection-track-stats.https.html index 3809530b74cac3..9ad679272b57e4 100644 --- a/webrtc/RTCPeerConnection-track-stats.https.html +++ b/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 ' + diff --git a/webrtc/RTCRtpReceiver-getStats.https.html b/webrtc/RTCRtpReceiver-getStats.https.html index 05ca9f3c90b652..7cd598393708b5 100644 --- a/webrtc/RTCRtpReceiver-getStats.https.html +++ b/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');