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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
| commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
| tree | f435a8308119effd964b339f76abb83a57c29483 /testing/web-platform/meta/webrtc-extensions | |
| parent | Initial commit. (diff) | |
| download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip | |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/meta/webrtc-extensions')
10 files changed, 86 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..13473feaa1 --- /dev/null +++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpParameters-codec.html.ini @@ -0,0 +1,55 @@ +[RTCRtpParameters-codec.html] + expected: ERROR + [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 + + [Creating an audio sender with addTransceiver and non-existing codec type should throw OperationError] + expected: FAIL + + [Creating a video sender with addTransceiver and non-existing codec type should throw OperationError] + expected: FAIL + + [Stats output-rtp should match the selected codec in simulcast usecase on a video sender] + expected: FAIL + + [Stats output-rtp should match the selected mixed codecs in simulcast usecase on a video sender] + expected: FAIL 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..a96b98ec88 --- /dev/null +++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html.ini @@ -0,0 +1,13 @@ +[RTCRtpReceiver-jitterBufferTarget-stats.html] + expected: + if (os == "android") and not debug: [OK, TIMEOUT] + [measure raising and lowering video jitterBufferTarget] + expected: + if (os == "win") and not debug and (processor == "x86"): [PASS, FAIL] + if (os == "android") and not debug: [PASS, FAIL, TIMEOUT] + if os == "linux": [PASS, FAIL] + + [measure raising and lowering audio jitterBufferTarget] + expected: + if (os == "android") and debug and swgl: [PASS, FAIL] + if (os == "android") and not debug: [PASS, FAIL, TIMEOUT] 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 |