Bug 1728628 [wpt PR 30287] - Update wpt metadata, a=testonly

wpt-pr: 30287 wpt-type: metadata
PLSysSec · Sep 5, 2021 · 1c2fca1 · 1c2fca1
1 parent 0281b06
commit 1c2fca1
Show file tree

Hide file tree

Showing 4 changed files with 110 additions and 5 deletions.
diff --git a/...web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-captureTimestamp.html.ini b/...web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-captureTimestamp.html.ini
@@ -2,13 +2,36 @@
   expected:
     if (os == "linux") and not debug and not webrender and (processor == "x86_64"): [OK, TIMEOUT]
     if (os == "linux") and debug and not webrender: [OK, ERROR]
-    if (os == "linux") and webrender and debug and not swgl: ["OK", "TIMEOUT"]
+    if (os == "linux") and webrender and debug and not swgl: [OK, TIMEOUT]
+    if (os == "win") and not debug and (processor == "x86"): [OK, TIMEOUT]
+    if (os == "linux") and swgl: [OK, TIMEOUT]
   [Audio and video RTCRtpSynchronizationSource.captureTimestamp are comparable]
-    expected: FAIL
+    expected:
+      if (processor == "x86") and not debug: [FAIL, NOTRUN]
+      FAIL
 
   [[audio\] getSynchronizationSources() should contain captureTimestamp if absolute capture time RTP header extension is negotiated]
-    expected: FAIL
+    expected:
+      if (processor == "x86") and not debug: [FAIL, NOTRUN]
+      FAIL
 
   [[video\] getSynchronizationSources() should contain captureTimestamp if absolute capture time RTP header extension is negotiated]
-    expected: FAIL
+    expected:
+      if (processor == "x86") and not debug: [FAIL, NOTRUN]
+      FAIL
 
+  [[audio\] getSynchronizationSources() should not contain captureTimestamp if absolute capture time RTP header extension is not offered]
+    expected:
+      if (processor == "x86") and not debug: [PASS, TIMEOUT]
+
+  [[audio\] getSynchronizationSources() should not contain captureTimestamp if absolute capture time RTP header extension is offered, but not answered]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[video\] getSynchronizationSources() should not contain captureTimestamp if absolute capture time RTP header extension is not offered]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[video\] getSynchronizationSources() should not contain captureTimestamp if absolute capture time RTP header extension is offered, but not answered]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-bufferedAmount.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-bufferedAmount.html.ini
@@ -1,98 +1,123 @@
 [RTCDataChannel-bufferedAmount.html]
   expected:
     if (os == "mac") and not debug: [OK, TIMEOUT]
+    if (os == "win") and not debug and (processor == "x86_64"): [OK, TIMEOUT]
   [negotiated datachannel bufferedAmount should increase to byte length of encodedunicode string sent]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedAmount initial value should be 0 for both peers]
     expected:
       if (os == "mac") and not debug: [PASS, TIMEOUT]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, TIMEOUT]
 
   [negotiated datachannel bufferedamount returns the same amount if no more data is]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedAmount should not decrease immediately after initiating closure]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedamountlow event fires only once after multiple consecutive send() calls]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedamountlow event fires after each sent message]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedAmount should increase to size of blob sent]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN, TIMEOUT]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedAmount should not decrease after closing the peer connection]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedAmount should not decrease immediately after initiating closure]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedamount is data.length on send(data)]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedAmount should not decrease after closing the peer connection]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedAmount should increase to byte length of buffer sent]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedAmount should increase to byte length of encodedunicode string sent]
     expected:
       if (os == "mac") and not debug: [PASS, TIMEOUT, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedAmount should increase to size of blob sent]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedAmount should increase to byte length of buffer sent]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN, TIMEOUT]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedamount returns the same amount if no more data is]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedamount is data.length on send(data)]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedamountlow event fires after send() is complete]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedamountlow event fires only once after multiple consecutive send() calls]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedAmount should increase by byte length for each message sent]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedAmount initial value should be 0 for both peers]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedamountlow event fires after send() is complete]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [negotiated datachannel bufferedamountlow event fires after each sent message]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
 
   [datachannel bufferedAmount should increase by byte length for each message sent]
     expected:
       if (os == "mac") and not debug: [PASS, NOTRUN]
+      if (os == "win") and not debug and (processor == "x86_64"): [PASS, NOTRUN]
diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-helper-test.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-helper-test.html.ini
@@ -4,3 +4,6 @@
   [Setting up a connection using helpers and defaults should work]
     expected:
       if (os == "mac") and not debug: [PASS, TIMEOUT]
+
+  [A test video track transmits at least 40 Kbits/sec of data at 480x360 size]
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini
@@ -2,7 +2,61 @@
   expected:
     if (os == "linux") and not debug and webrender and not fission: [OK, ERROR, CRASH]
     if (os == "linux") and not debug and not webrender: [OK, ERROR, CRASH]
+    if (os == "win") and not debug and (processor == "x86"): [OK, TIMEOUT]
   [[audio-only\] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean]
     bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1525394
-    expected: FAIL
+    expected:
+      if (processor == "x86") and not debug: [FAIL, NOTRUN]
+      FAIL
 
+  [[audio\] getSynchronizationSources() eventually returns a non-empty list]
+    expected:
+      if (processor == "x86") and not debug: [PASS, TIMEOUT]
+
+  [[audio\] RTCRtpSynchronizationSource.timestamp is a number]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[audio\] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1\]]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[audio\] getSynchronizationSources() does not contain SSRCs older than 10 seconds]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[audio\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[audio\] RTCRtpSynchronizationSource.source is a number]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[video\] getSynchronizationSources() eventually returns a non-empty list]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[video\] RTCRtpSynchronizationSource.timestamp is a number]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[video\] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1\]]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[video\] getSynchronizationSources() does not contain SSRCs older than 10 seconds]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[video\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[video\] RTCRtpSynchronizationSource.source is a number]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]
+
+  [[audio-only\] RTCRtpSynchronizationSource.audioLevel is a number [0, 1\]]
+    expected:
+      if (processor == "x86") and not debug: [PASS, NOTRUN]