summaryrefslogtreecommitdiffstats
path: root/testing/web-platform/meta/webrtc-extensions
diff options
context:
space:
mode:
Diffstat (limited to 'testing/web-platform/meta/webrtc-extensions')
-rw-r--r--testing/web-platform/meta/webrtc-extensions/RTCOAuthCredential.html.ini2
-rw-r--r--testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-adaptivePtime.html.ini2
-rw-r--r--testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-codec.html.ini48
-rw-r--r--testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-maxFramerate.html.ini3
-rw-r--r--testing/web-platform/meta/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html.ini6
-rw-r--r--testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-captureTimestamp.html.ini4
-rw-r--r--testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html.ini3
-rw-r--r--testing/web-platform/meta/webrtc-extensions/RTCRtpTransceiver-headerExtensionControl.html.ini2
-rw-r--r--testing/web-platform/meta/webrtc-extensions/__dir__.ini1
-rw-r--r--testing/web-platform/meta/webrtc-extensions/transfer-datachannel-service-worker.https.html.ini2
-rw-r--r--testing/web-platform/meta/webrtc-extensions/transfer-datachannel.html.ini2
11 files changed, 75 insertions, 0 deletions
diff --git a/testing/web-platform/meta/webrtc-extensions/RTCOAuthCredential.html.ini b/testing/web-platform/meta/webrtc-extensions/RTCOAuthCredential.html.ini
new file mode 100644
index 0000000000..8f1c728089
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/RTCOAuthCredential.html.ini
@@ -0,0 +1,2 @@
+[RTCOAuthCredential.html]
+ disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1247616
diff --git a/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-adaptivePtime.html.ini b/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-adaptivePtime.html.ini
new file mode 100644
index 0000000000..b4f005ae5e
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-adaptivePtime.html.ini
@@ -0,0 +1,2 @@
+[RTCRtpParameters-adaptivePtime.html]
+ disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1733647
diff --git a/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-codec.html.ini b/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-codec.html.ini
new file mode 100644
index 0000000000..ed1f0cc257
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-codec.html.ini
@@ -0,0 +1,48 @@
+[RTCRtpParameters-codec.html]
+ [Creating an audio sender with addTransceiver and codec should work]
+ expected: FAIL
+
+ [Creating a video sender with addTransceiver and codec should work]
+ expected: FAIL
+
+ [Setting codec on an audio sender with setParameters should work]
+ expected: FAIL
+
+ [Setting codec on a video sender with setParameters should work]
+ expected: FAIL
+
+ [Creating an audio sender with addTransceiver and non-existing codec should throw OperationError]
+ expected: FAIL
+
+ [Creating a video sender with addTransceiver and non-existing codec should throw OperationError]
+ expected: FAIL
+
+ [Setting a non-existing codec on an audio sender with setParameters should throw InvalidModificationError]
+ expected: FAIL
+
+ [Setting a non-existing codec on a video sender with setParameters should throw InvalidModificationError]
+ expected: FAIL
+
+ [Setting a non-preferred codec on an audio sender with setParameters should throw InvalidModificationError]
+ expected: FAIL
+
+ [Setting a non-preferred codec on a video sender with setParameters should throw InvalidModificationError]
+ expected: FAIL
+
+ [Setting a non-negotiated codec on an audio sender with setParameters should throw InvalidModificationError]
+ expected: FAIL
+
+ [Setting a non-negotiated codec on a video sender with setParameters should throw InvalidModificationError]
+ expected: FAIL
+
+ [Codec should be undefined after negotiating away the currently set codec on an audio sender]
+ expected: FAIL
+
+ [Codec should be undefined after negotiating away the currently set codec on a video sender]
+ expected: FAIL
+
+ [Stats output-rtp should match the selected codec in non-simulcast usecase on an audio sender]
+ expected: FAIL
+
+ [Stats output-rtp should match the selected codec in non-simulcast usecase on a video sender]
+ expected: FAIL
diff --git a/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-maxFramerate.html.ini b/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-maxFramerate.html.ini
new file mode 100644
index 0000000000..4a86cadbf4
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-maxFramerate.html.ini
@@ -0,0 +1,3 @@
+[RTCRtpParameters-maxFramerate.html]
+ expected:
+ if (os == "android") and fission: [OK, TIMEOUT]
diff --git a/testing/web-platform/meta/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html.ini b/testing/web-platform/meta/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html.ini
new file mode 100644
index 0000000000..fb35a55895
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html.ini
@@ -0,0 +1,6 @@
+[RTCRtpReceiver-jitterBufferTarget-stats.html]
+
+ [measure raising and lowering video jitterBufferTarget]
+ expected:
+ if (os == "linux"): [FAIL, PASS]
+
diff --git a/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-captureTimestamp.html.ini b/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-captureTimestamp.html.ini
new file mode 100644
index 0000000000..3024f3f627
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-captureTimestamp.html.ini
@@ -0,0 +1,4 @@
+[RTCRtpSynchronizationSource-captureTimestamp.html]
+ disabled: true
+ bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1733653
+
diff --git a/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html.ini b/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html.ini
new file mode 100644
index 0000000000..3fb6aa2f71
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html.ini
@@ -0,0 +1,3 @@
+[RTCRtpSynchronizationSource-senderCaptureTimeOffset.html]
+ disabled: true
+ bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1733653
diff --git a/testing/web-platform/meta/webrtc-extensions/RTCRtpTransceiver-headerExtensionControl.html.ini b/testing/web-platform/meta/webrtc-extensions/RTCRtpTransceiver-headerExtensionControl.html.ini
new file mode 100644
index 0000000000..f18573b4b0
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpTransceiver-headerExtensionControl.html.ini
@@ -0,0 +1,2 @@
+[RTCRtpTransceiver-headerExtensionControl.html]
+ disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1733654
diff --git a/testing/web-platform/meta/webrtc-extensions/__dir__.ini b/testing/web-platform/meta/webrtc-extensions/__dir__.ini
new file mode 100644
index 0000000000..9703cbb378
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/__dir__.ini
@@ -0,0 +1 @@
+leak-threshold: [default:3020800, rdd:51200]
diff --git a/testing/web-platform/meta/webrtc-extensions/transfer-datachannel-service-worker.https.html.ini b/testing/web-platform/meta/webrtc-extensions/transfer-datachannel-service-worker.https.html.ini
new file mode 100644
index 0000000000..c635355a97
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/transfer-datachannel-service-worker.https.html.ini
@@ -0,0 +1,2 @@
+[transfer-datachannel-service-worker.https.html]
+ disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1209163
diff --git a/testing/web-platform/meta/webrtc-extensions/transfer-datachannel.html.ini b/testing/web-platform/meta/webrtc-extensions/transfer-datachannel.html.ini
new file mode 100644
index 0000000000..3134a1a0e1
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-extensions/transfer-datachannel.html.ini
@@ -0,0 +1,2 @@
+[transfer-datachannel.html]
+ disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1209163