RTCRtpSender.replaceTrack added behind flag.
Wires up replaceTrack[1] from the content layer implementation[2],
exposing it in JavaScript behind RuntimeEnabled experimental feature
"RTCRtpSenderReplaceTrack".
The meat of this CL are the tests. This includes making sure the track
is set asynchronously (tested with WPT) and that replacing a video
track causes different video to be sent (tested with browser_tests
instead of WPT due to bug https://crbug.com/793808).
Design doc:
https://docs.google.com/document/d/1bpsYLC35cHAJnlnmiBJ3bhUQXfUBgKbTv7A6nkEs6OA/edit?usp=sharing
[1] https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-replacetrack
[2] https://chromium-review.googlesource.com/c/chromium/src/+/833870/
Bug: 790007
Change-Id: If8c3499c78fe664a496c5d90cfbfa7b48c1c036f
Reviewed-on: https://chromium-review.googlesource.com/810765
Reviewed-by: Philip Jägenstedt <foolip@chromium.org>
Reviewed-by: Harald Alvestrand <hta@chromium.org>
Reviewed-by: Taylor Brandstetter <deadbeef@chromium.org>
Commit-Queue: Henrik Boström <hbos@chromium.org>
Cr-Commit-Position: refs/heads/master@{#528071}
diff --git a/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html b/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html
new file mode 100644
index 0000000..8a40f9a
--- /dev/null
+++ b/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html
@@ -0,0 +1,139 @@
+<!doctype html>
+<meta charset=utf-8>
+<title>RTCPeerConnection.prototype.setRemoteDescription - replaceTrack</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script src="RTCPeerConnection-helper.js"></script>
+<script>
+ 'use strict';
+
+ // The following helper functions are called from RTCPeerConnection-helper.js:
+ // getUserMediaTracksAndStreams
+
+ async_test(t => {
+ const caller = new RTCPeerConnection();
+ return getUserMediaTracksAndStreams(2)
+ .then(t.step_func(([tracks, streams]) => {
+ const sender = caller.addTrack(tracks[0]);
+ return sender.replaceTrack(tracks[1])
+ .then(t.step_func(() => {
+ assert_equals(sender.track, tracks[1]);
+ t.done();
+ }));
+ }))
+ .catch(t.step_func(reason => {
+ assert_unreached(reason);
+ }));
+ }, 'replaceTrack() sets the track attribute to a new track.');
+
+ async_test(t => {
+ const caller = new RTCPeerConnection();
+ return getUserMediaTracksAndStreams(1)
+ .then(t.step_func(([tracks, streams]) => {
+ const sender = caller.addTrack(tracks[0]);
+ return sender.replaceTrack(null)
+ .then(t.step_func(() => {
+ assert_equals(sender.track, null);
+ t.done();
+ }));
+ }))
+ .catch(t.step_func(reason => {
+ assert_unreached(reason);
+ }));
+ }, 'replaceTrack() sets the track attribute to null.');
+
+ async_test(t => {
+ const caller = new RTCPeerConnection();
+ return getUserMediaTracksAndStreams(2)
+ .then(t.step_func(([tracks, streams]) => {
+ const sender = caller.addTrack(tracks[0]);
+ assert_equals(sender.track, tracks[0]);
+ sender.replaceTrack(tracks[1]);
+ // replaceTrack() is asynchronous, there should be no synchronously
+ // observable effects.
+ assert_equals(sender.track, tracks[0]);
+ t.done();
+ }))
+ .catch(t.step_func(reason => {
+ assert_unreached(reason);
+ }));
+ }, 'replaceTrack() does not set the track synchronously.');
+
+ async_test(t => {
+ const expectedException = 'InvalidStateError';
+ const caller = new RTCPeerConnection();
+ return getUserMediaTracksAndStreams(2)
+ .then(t.step_func(([tracks, streams]) => {
+ const sender = caller.addTrack(tracks[0]);
+ caller.close();
+ return sender.replaceTrack(tracks[1])
+ .then(t.step_func(() => {
+ assert_unreached('Expected replaceTrack() to be rejected with ' +
+ expectedException + ' but the promise was resolved.');
+ }),
+ t.step_func(e => {
+ assert_equals(e.name, expectedException);
+ t.done();
+ }));
+ }))
+ .catch(t.step_func(reason => {
+ assert_unreached(reason);
+ }));
+ }, 'replaceTrack() rejects when the peer connection is closed.');
+
+ async_test(t => {
+ const expectedException = 'InvalidModificationError';
+ const caller = new RTCPeerConnection();
+ return getUserMediaTracksAndStreams(2)
+ .then(t.step_func(([tracks, streams]) => {
+ const sender = caller.addTrack(tracks[0]);
+ caller.removeTrack(sender);
+ // replaceTrack() should fail because the sender should be inactive after
+ // removeTrack().
+ return sender.replaceTrack(tracks[1])
+ .then(t.step_func(() => {
+ assert_unreached('Expected replaceTrack() to be rejected with ' +
+ expectedException + ' but the promise was resolved.');
+ }),
+ t.step_func(e => {
+ assert_equals(e.name, expectedException);
+ t.done();
+ }));
+ }))
+ .catch(t.step_func(reason => {
+ assert_unreached(reason);
+ }));
+ }, 'replaceTrack() rejects when invoked after removeTrack().');
+
+ async_test(t => {
+ const expectedException = 'InvalidModificationError';
+ const caller = new RTCPeerConnection();
+ return getUserMediaTracksAndStreams(2)
+ .then(t.step_func(([tracks, streams]) => {
+ const sender = caller.addTrack(tracks[0]);
+ let p = sender.replaceTrack(tracks[1])
+ caller.removeTrack(sender);
+ // replaceTrack() should fail because it executes steps in parallel and
+ // queues a task to execute after removeTrack() has occurred. The sender
+ // should be inactive. If this can be racy, update or remove the test.
+ // https://github.com/w3c/webrtc-pc/issues/1728
+ return p.then(t.step_func(() => {
+ assert_unreached('Expected replaceTrack() to be rejected with ' +
+ expectedException + ' but the promise was resolved.');
+ }),
+ t.step_func(e => {
+ assert_equals(e.name, expectedException);
+ t.done();
+ }));
+ }))
+ .catch(t.step_func(reason => {
+ assert_unreached(reason);
+ }));
+ }, 'replaceTrack() rejects after a subsequent removeTrack().');
+
+ // TODO(hbos): Verify that replaceTrack() changes what media is received on
+ // the remote end of two connected peer connections. For video tracks, this
+ // requires Chromium's video tag to update on receiving frames when running
+ // content_shell. https://crbug.com/793808
+
+</script>
