Adding upstream version 125.0.1.upstream/125.0.1

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

1 files changed, 21 insertions, 1 deletions

diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini
index 6b8799454b..bb03b74d75 100644
--- a/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini
+++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini
@@ -1,4 +1,24 @@
 [RTCRtpReceiver-getSynchronizationSources.https.html]
+  expected:
+    if (os == "android") and not debug: [OK, TIMEOUT]
   [[audio-only\] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean]
     bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1525394
-    expected: FAIL
+    expected:
+      if (os == "android") and not debug: [FAIL, NOTRUN]
+      FAIL
+
+  [[video\] getSynchronizationSources() does not contain SSRCs older than 10 seconds]
+    expected:
+      if (os == "android") and not debug: [PASS, TIMEOUT]
+
+  [[video\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()]
+    expected:
+      if (os == "android") and not debug: [PASS, NOTRUN]
+
+  [[video\] RTCRtpSynchronizationSource.source is a number]
+    expected:
+      if (os == "android") and not debug: [PASS, NOTRUN]
+
+  [[audio-only\] RTCRtpSynchronizationSource.audioLevel is a number [0, 1\]]
+    expected:
+      if (os == "android") and not debug: [PASS, NOTRUN]