Adding upstream version 124.0.1.upstream/124.0.1

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

23 files changed, 174 insertions, 0 deletions

diff --git a/testing/web-platform/meta/webrtc/protocol/__dir__.ini b/testing/web-platform/meta/webrtc/protocol/__dir__.ini
new file mode 100644
index 0000000000..c6a51b9705
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/__dir__.ini
@@ -0,0 +1,2 @@
+lsan-allowed: [Create, CreateNullDecoderModule, IPC::Channel::Channel, MakeRefPtr, MakeUnique, NewPage, NewSegment, PLDHashTable::Add, PLDHashTable::ChangeTable, PLDHashTable::MakeEntryHandle, Realloc, RevocableStore::RevocableStore, allocate, already_AddRefed, maybe_pod_malloc, mozilla::FFmpegDecoderModule, mozilla::KnowsCompositorVideo::TryCreateForIdentifier, mozilla::detail::UniqueSelector, mozilla::ipc::IProtocol::ActorConnected, mozilla::ipc::MessageChannel::Open, mozilla::layers::BufferTextureData::CreateInternal, mozilla::layers::ImageContainer::CreatePlanarYCbCrImage, mozilla::layers::ImageContainer::EnsureRecycleAllocatorForRDD, mozilla::layers::TextureClient::CreateIPDLActor, mozilla::layers::TextureClientRecycleAllocator::CreateOrRecycle, mozilla::layers::VideoBridgeChild::Open, sctp_add_vtag_to_timewait, sctp_hashinit_flags]
+leak-threshold: [default:3020800, rdd:51200]
diff --git a/testing/web-platform/meta/webrtc/protocol/additional-codecs.html.ini b/testing/web-platform/meta/webrtc/protocol/additional-codecs.html.ini
new file mode 100644
index 0000000000..a0f2f64f43
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/additional-codecs.html.ini
@@ -0,0 +1,3 @@
+[additional-codecs.html]
+  [Listing an additional codec in the answer causes it to be sent.]
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini b/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini
new file mode 100644
index 0000000000..8c84464872
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini
@@ -0,0 +1,23 @@
+[bundle.https.html]
+  expected:
+    if (os == "android") and debug and not swgl: [OK, TIMEOUT]
+    if (os == "win") and not debug and (processor == "x86"): [OK, CRASH]
+  [not negotiating BUNDLE creates two separate ice and dtls transports]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996
+    expected: FAIL
+
+  [bundles on the first transport and closes the second]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1805480
+    expected:
+      if (os == "android") and debug and not swgl: [FAIL, TIMEOUT]
+      FAIL
+
+  [max-bundle with an offer without bundle only negotiates the first m-line]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1805484
+    expected:
+      if (os == "android") and debug and not swgl: [FAIL, NOTRUN]
+      FAIL
+
+  [sRD(offer) works with no transport attributes in a bundle-only m-section]
+    expected:
+      if (os == "android") and debug and not swgl: [PASS, NOTRUN]
diff --git a/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini b/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini
new file mode 100644
index 0000000000..297b54b1f8
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini
@@ -0,0 +1,18 @@
+[candidate-exchange.https.html]
+  expected:
+    if (os == "linux") and not debug and fission: [OK, CRASH]
+  [Adding only caller -> callee candidates gives a connection]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
+
+  [Adding only callee -> caller candidates gives a connection]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
+
+  [Explicit offer/answer exchange gives a connection]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
+
+  [Adding callee -> caller candidates from end-of-candidates gives a connection]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini b/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini
new file mode 100644
index 0000000000..116d12f4e4
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini
@@ -0,0 +1,24 @@
+[crypto-suite.https.html]
+  [srtpCipher is acceptable on video-only]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
+
+  [srtpCipher is acceptable on data-only]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
+
+  [dtlsCipher is acceptable on video-only]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
+
+  [dtlsCipher is acceptable on data-only]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
+
+  [tlsVersion is acceptable on video-only]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
+
+  [tlsVersion is acceptable on data-only]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/dtls-certificates.html.ini b/testing/web-platform/meta/webrtc/protocol/dtls-certificates.html.ini
new file mode 100644
index 0000000000..3e512c9e93
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/dtls-certificates.html.ini
@@ -0,0 +1,6 @@
+[dtls-certificates.html]
+  expected:
+    if (os == "android") and not debug: [OK, TIMEOUT]
+  [RTCPeerConnection establishes using rsa and rsa certificates]
+    expected:
+      if (os == "android") and not debug: [PASS, TIMEOUT]
diff --git a/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini b/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini
new file mode 100644
index 0000000000..34cafce8e1
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini
@@ -0,0 +1,3 @@
+[dtls-fingerprint-validation.html]
+  expected:
+    if tsan: CRASH
diff --git a/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini b/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini
new file mode 100644
index 0000000000..3ad3443d9b
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini
@@ -0,0 +1,16 @@
+[dtls-setup.https.html]
+  [PC with setup=actpass should have a dtlsRole of client]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723
+    expected: FAIL
+
+  [PC with setup=active should have a dtlsRole of server]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723
+    expected: FAIL
+
+  [PC with setup=passive should have a dtlsRole of client]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723
+    expected: FAIL
+
+  [dtlsRole is `unknown` before negotiation of the DTLS handshake]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini b/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini
new file mode 100644
index 0000000000..2f72b22a32
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini
@@ -0,0 +1,2 @@
+[h264-profile-levels.https.html]
+  disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1396922
diff --git a/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini b/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini
new file mode 100644
index 0000000000..9702bc1803
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini
@@ -0,0 +1,2 @@
+[handover-datachannel.html]
+  disabled: https://github.com/web-platform-tests/wpt/issues/37561 
diff --git a/testing/web-platform/meta/webrtc/protocol/handover.html.ini b/testing/web-platform/meta/webrtc/protocol/handover.html.ini
new file mode 100644
index 0000000000..b8feb75485
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/handover.html.ini
@@ -0,0 +1,2 @@
+[handover.html]
+  disabled: https://github.com/web-platform-tests/wpt/issues/37561 
diff --git a/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini b/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini
new file mode 100644
index 0000000000..9b13a9e695
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini
@@ -0,0 +1,4 @@
+[ice-state.https.html]
+  [PC should enter disconnected state when a failing candidate is sent]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1557053
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini b/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini
new file mode 100644
index 0000000000..1dcc567dbc
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini
@@ -0,0 +1,8 @@
+[ice-ufragpwd.html]
+  [setRemoteDescription with a ice-ufrag containing a non-ice-char fails]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1617686
+    expected: FAIL
+
+  [setRemoteDescription with a ice-pwd containing a non-ice-char fails]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1617686
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini
new file mode 100644
index 0000000000..f8b32255b3
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini
@@ -0,0 +1,5 @@
+[rtp-clockrate.html]
+  [video rtp timestamps increase by approximately 90000 per second]
+    expected:
+      if (os == "win") and not debug and (processor == "x86"): [PASS, FAIL]
+      if (os == "mac") and not debug: [PASS, FAIL]
diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini
new file mode 100644
index 0000000000..186f43bc00
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini
@@ -0,0 +1,10 @@
+[rtp-demuxing.html]
+  bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1818283
+  expected: [OK, TIMEOUT]
+  [Can demux two video tracks with different payload types on a bundled connection]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1396922
+    expected: FAIL
+
+  [Can demux two video tracks with the same payload type on an unbundled connection]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1818283
+    expected: [PASS, TIMEOUT]
diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini
new file mode 100644
index 0000000000..44bb7ecbae
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini
@@ -0,0 +1,4 @@
+[rtp-extension-support.html]
+  [RTP header extension urn:3gpp:video-orientation is present in offer]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1340372
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-headerextensions.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-headerextensions.html.ini
new file mode 100644
index 0000000000..af9dc6374b
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/rtp-headerextensions.html.ini
@@ -0,0 +1,12 @@
+[rtp-headerextensions.html]
+  [Video orientation header extension is supported.]
+    expected: FAIL
+
+  [Negotiates the subset of supported extensions offered]
+    expected: FAIL
+
+  [Supports header extensions with id=15]
+    expected: FAIL
+
+  [Supports two-byte header extensions]
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini
new file mode 100644
index 0000000000..59d1862d17
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini
@@ -0,0 +1,4 @@
+[rtp-payloadtypes.html]
+  [setRemoteDescription with a codec in the range 64-95 throws an InvalidAccessError]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1806181
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/simulcast-offer.html.ini b/testing/web-platform/meta/webrtc/protocol/simulcast-offer.html.ini
new file mode 100644
index 0000000000..1422fe0bc8
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/simulcast-offer.html.ini
@@ -0,0 +1,3 @@
+[simulcast-offer.html]
+  expected:
+    if (os == "android") and fission: [OK, TIMEOUT]
diff --git a/testing/web-platform/meta/webrtc/protocol/transceiver-mline-recycling.html.ini b/testing/web-platform/meta/webrtc/protocol/transceiver-mline-recycling.html.ini
new file mode 100644
index 0000000000..4ac0c264e1
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/transceiver-mline-recycling.html.ini
@@ -0,0 +1,6 @@
+[transceiver-mline-recycling.html]
+  [Reuses m-lines in local negotiation]
+    expected: FAIL
+
+  [Reuses m-lines in remote negotiation]
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini b/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini
new file mode 100644
index 0000000000..9a216b2119
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini
@@ -0,0 +1,4 @@
+[unknown-mediatypes.html]
+  [Unknown media types are rejected with the port set to 0]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1806185
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini b/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini
new file mode 100644
index 0000000000..659a322d55
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini
@@ -0,0 +1,9 @@
+[video-codecs.https.html]
+  max-asserts: 3
+  [H.264 and VP8 should be supported in initial offer]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1534688
+    expected: FAIL
+
+  [H.264 and VP8 should be negotiated after handshake]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1534687
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini b/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini
new file mode 100644
index 0000000000..812d1ea704
--- /dev/null
+++ b/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini
@@ -0,0 +1,4 @@
+[vp8-fmtp.html]
+  [setRemoteDescription parses max-fr and max-fs fmtp parameters]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531464
+    expected: FAIL