Merging upstream version 127.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

7 files changed, 30 insertions, 10 deletions

diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-send-close.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-send-close.html.ini
index e9f2be5f4f..63a12be1c6 100644
--- a/testing/web-platform/meta/webrtc/RTCDataChannel-send-close.html.ini
+++ b/testing/web-platform/meta/webrtc/RTCDataChannel-send-close.html.ini
@@ -1,7 +1,6 @@
 [RTCDataChannel-send-close.html]
   expected:
-    if (os == "win") and (processor == "x86"): CRASH
-    if os == "android": CRASH
+    if os == "android": [CRASH, TIMEOUT]
     [TIMEOUT, OK]
   [Datachannel should be able to send and receive all string messages on close]
     expected: [FAIL, TIMEOUT, NOTRUN]
diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini
index 7fc41ec7d8..ff8279db28 100644
--- a/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini
+++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini
@@ -1,4 +1,6 @@
 [RTCPeerConnection-iceConnectionState.https.html]
+  expected:
+    if tsan: CRASH
   [iceConnectionState changes at the right time, with bundle policy max-bundle]
     bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
     expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html.ini
new file mode 100644
index 0000000000..fa8ca81b19
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html.ini
@@ -0,0 +1,3 @@
+[RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html]
+  [measure raising and lowering audio jitterBufferTarget]
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-video-jitterBufferTarget-stats.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-video-jitterBufferTarget-stats.html.ini
new file mode 100644
index 0000000000..511e0d93b3
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-video-jitterBufferTarget-stats.html.ini
@@ -0,0 +1,3 @@
+[RTCRtpReceiver-video-jitterBufferTarget-stats.html]
+  [measure raising and lowering video jitterBufferTarget]
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini b/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini
index 6a26d4498c..37d9821fde 100644
--- a/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini
+++ b/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini
@@ -215,26 +215,29 @@
   [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "address" with the proper type]
     expected: FAIL
 
-  [RTCRtpTransceiver interface: operation setCodecPreferences(sequence<RTCRtpCodecCapability>)]
+  [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "binaryType" with the proper type]
     expected: FAIL
 
-  [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "setCodecPreferences(sequence<RTCRtpCodecCapability>)" with the proper type]
+  [RTCIceCandidate interface: attribute relayProtocol]
     expected: FAIL
 
-  [RTCRtpTransceiver interface: calling setCodecPreferences(sequence<RTCRtpCodecCapability>) on new RTCPeerConnection().addTransceiver('audio') with too few arguments must throw TypeError]
+  [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relayProtocol" with the proper type]
     expected: FAIL
 
-  [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "binaryType" with the proper type]
+  [RTCIceCandidate interface: attribute url]
     expected: FAIL
 
-  [RTCIceCandidate interface: attribute relayProtocol]
+  [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "url" with the proper type]
     expected: FAIL
 
-  [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relayProtocol" with the proper type]
+  [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "jitterBufferTarget" with the proper type]
     expected: FAIL
 
-  [RTCIceCandidate interface: attribute url]
+  [RTCRtpTransceiver interface: operation setCodecPreferences(sequence<RTCRtpCodec>)]
     expected: FAIL
 
-  [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "url" with the proper type]
+  [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "setCodecPreferences(sequence<RTCRtpCodec>)" with the proper type]
+    expected: FAIL
+
+  [RTCRtpTransceiver interface: calling setCodecPreferences(sequence<RTCRtpCodec>) on new RTCPeerConnection().addTransceiver('audio') with too few arguments must throw TypeError]
     expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/legacy/simplecall_callbacks.https.html.ini b/testing/web-platform/meta/webrtc/legacy/simplecall_callbacks.https.html.ini
new file mode 100644
index 0000000000..f640335196
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/legacy/simplecall_callbacks.https.html.ini
@@ -0,0 +1,6 @@
+[simplecall_callbacks.https.html]
+  expected:
+    if (os == "mac") and not debug: [OK, TIMEOUT]
+  [Can set up a basic WebRTC call.]
+    expected:
+      if (os == "mac") and not debug: [PASS, TIMEOUT]
diff --git a/testing/web-platform/meta/webrtc/simulcast/setParameters-active.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/setParameters-active.https.html.ini
new file mode 100644
index 0000000000..9a1ece0772
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/simulcast/setParameters-active.https.html.ini
@@ -0,0 +1,4 @@
+[setParameters-active.https.html]
+  [Simulcast setParameters active=false stops sending frames]
+    expected:
+      if (os == "mac") and not debug: [PASS, FAIL]