Adding upstream version 124.0.1.upstream/124.0.1

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

5 files changed, 479 insertions, 0 deletions

diff --git a/testing/web-platform/meta/webrtc-stats/getStats-remote-candidate-address.html.ini b/testing/web-platform/meta/webrtc-stats/getStats-remote-candidate-address.html.ini
new file mode 100644
index 0000000000..6474cce833
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-stats/getStats-remote-candidate-address.html.ini
@@ -0,0 +1,6 @@
+[getStats-remote-candidate-address.html]
+  expected:
+    if os == "mac": [OK, CRASH]
+  [Do not expose in stats remote addresses that are not known to be already exposed to JS]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1534701
+    expected: FAIL
diff --git a/testing/web-platform/meta/webrtc-stats/hardware-capability-stats.https.html.ini b/testing/web-platform/meta/webrtc-stats/hardware-capability-stats.https.html.ini
new file mode 100644
index 0000000000..370ba59fbc
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-stats/hardware-capability-stats.https.html.ini
@@ -0,0 +1,2 @@
+[hardware-capability-stats.https.html]
+  disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1804977
diff --git a/testing/web-platform/meta/webrtc-stats/outbound-rtp.https.html.ini b/testing/web-platform/meta/webrtc-stats/outbound-rtp.https.html.ini
new file mode 100644
index 0000000000..4a01d8cf0c
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-stats/outbound-rtp.https.html.ini
@@ -0,0 +1,5 @@
+[outbound-rtp.https.html]
+  [setting an encoding to false is reflected in outbound-rtp stats]
+    bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1813848
+    expected: FAIL
+
diff --git a/testing/web-platform/meta/webrtc-stats/rtp-stats-creation.html.ini b/testing/web-platform/meta/webrtc-stats/rtp-stats-creation.html.ini
new file mode 100644
index 0000000000..eb7656de24
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-stats/rtp-stats-creation.html.ini
@@ -0,0 +1,22 @@
+[rtp-stats-creation.html]
+  expected:
+    if (os == "win") and debug and not swgl: [OK, TIMEOUT]
+    if (os == "win") and not debug and (processor == "x86"): TIMEOUT
+    if os == "mac": [OK, TIMEOUT]
+  [No RTCInboundRtpStreamStats exist until packets have been received]
+    expected:
+      if (os == "win") and debug and swgl: [PASS, FAIL]
+      if (os == "win") and debug and not swgl: [PASS, FAIL, TIMEOUT]
+      if (os == "win") and not debug and (processor == "x86"): FAIL
+      if (os == "mac") and debug: [PASS, TIMEOUT]
+      if (os == "mac") and not debug: [PASS, FAIL, NOTRUN]
+
+  [RTCAudioPlayoutStats should be present]
+    expected:
+      if (os == "win") and not debug and (processor == "x86"): TIMEOUT
+      if (os == "mac") and not debug: [FAIL, TIMEOUT, NOTRUN]
+      FAIL
+
+  [No RTCOutboundRtpStreamStats exist until packets have been sent]
+    expected:
+      if (os == "mac") and not debug: [PASS, FAIL, TIMEOUT]
diff --git a/testing/web-platform/meta/webrtc-stats/supported-stats.https.html.ini b/testing/web-platform/meta/webrtc-stats/supported-stats.https.html.ini
new file mode 100644
index 0000000000..154afc059f
--- /dev/null
+++ b/testing/web-platform/meta/webrtc-stats/supported-stats.https.html.ini
@@ -0,0 +1,444 @@
+[supported-stats.https.html]
+  expected:
+    if (os == "android") and not debug: [OK, TIMEOUT]
+  [inbound-rtp's mid]
+    expected: FAIL
+
+  [inbound-rtp's keyFramesDecoded]
+    expected: FAIL
+
+  [inbound-rtp's qpSum]
+    expected:
+      if release_or_beta and not (os == "linux"): PASS
+      FAIL
+
+  [inbound-rtp's pauseCount]
+    expected: FAIL
+
+  [inbound-rtp's totalPausesDuration]
+    expected: FAIL
+
+  [inbound-rtp's freezeCount]
+    expected: FAIL
+
+  [inbound-rtp's totalFreezesDuration]
+    expected: FAIL
+
+  [inbound-rtp's estimatedPlayoutTimestamp]
+    expected: FAIL
+
+  [inbound-rtp's jitterBufferTargetDelay]
+    expected: FAIL
+
+  [inbound-rtp's jitterBufferMinimumDelay]
+    expected: FAIL
+
+  [inbound-rtp's decoderImplementation]
+    expected: FAIL
+
+  [inbound-rtp's playoutId]
+    expected: FAIL
+
+  [inbound-rtp's powerEfficientDecoder]
+    expected: FAIL
+
+  [inbound-rtp's framesAssembledFromMultiplePackets]
+    expected: FAIL
+
+  [inbound-rtp's totalAssemblyTime]
+    expected: FAIL
+
+  [inbound-rtp's transportId]
+    expected: FAIL
+
+  [outbound-rtp's mid]
+    expected: FAIL
+
+  [outbound-rtp's mediaSourceId]
+    expected: FAIL
+
+  [outbound-rtp's rid]
+    expected: PRECONDITION_FAILED
+
+  [outbound-rtp's targetBitrate]
+    expected: FAIL
+
+  [outbound-rtp's keyFramesEncoded]
+    expected: FAIL
+
+  [outbound-rtp's totalPacketSendDelay]
+    expected: FAIL
+
+  [outbound-rtp's qualityLimitationReason]
+    expected: FAIL
+
+  [outbound-rtp's qualityLimitationDurations]
+    expected: FAIL
+
+  [outbound-rtp's qualityLimitationResolutionChanges]
+    expected: FAIL
+
+  [outbound-rtp's encoderImplementation]
+    expected: FAIL
+
+  [outbound-rtp's powerEfficientEncoder]
+    expected: FAIL
+
+  [outbound-rtp's active]
+    expected: FAIL
+
+  [outbound-rtp's transportId]
+    expected: FAIL
+
+  [remote-inbound-rtp's transportId]
+    expected: FAIL
+
+  [remote-outbound-rtp's reportsSent]
+    expected: FAIL
+
+  [remote-outbound-rtp's roundTripTime]
+    expected: FAIL
+
+  [remote-outbound-rtp's totalRoundTripTime]
+    expected: FAIL
+
+  [remote-outbound-rtp's roundTripTimeMeasurements]
+    expected: FAIL
+
+  [remote-outbound-rtp's transportId]
+    expected: FAIL
+
+  [media-source's audioLevel]
+    expected: FAIL
+
+  [media-source's totalAudioEnergy]
+    expected: FAIL
+
+  [media-source's totalSamplesDuration]
+    expected: FAIL
+
+  [media-source's echoReturnLoss]
+    expected: PRECONDITION_FAILED
+
+  [media-source's echoReturnLossEnhancement]
+    expected: PRECONDITION_FAILED
+
+  [media-playout's synthesizedSamplesDuration]
+    expected: FAIL
+
+  [media-playout's synthesizedSamplesEvents]
+    expected: FAIL
+
+  [media-playout's totalSamplesDuration]
+    expected: FAIL
+
+  [media-playout's totalPlayoutDelay]
+    expected: FAIL
+
+  [media-playout's totalSamplesCount]
+    expected: FAIL
+
+  [media-playout's timestamp]
+    expected: FAIL
+
+  [media-playout's type]
+    expected: FAIL
+
+  [media-playout's id]
+    expected: FAIL
+
+  [transport's packetsSent]
+    expected: FAIL
+
+  [transport's packetsReceived]
+    expected: FAIL
+
+  [transport's bytesSent]
+    expected: FAIL
+
+  [transport's bytesReceived]
+    expected: FAIL
+
+  [transport's iceRole]
+    expected: FAIL
+
+  [transport's iceLocalUsernameFragment]
+    expected: FAIL
+
+  [transport's dtlsState]
+    expected: FAIL
+
+  [transport's iceState]
+    expected: FAIL
+
+  [transport's selectedCandidatePairId]
+    expected: FAIL
+
+  [transport's localCertificateId]
+    expected: FAIL
+
+  [transport's remoteCertificateId]
+    expected: FAIL
+
+  [transport's tlsVersion]
+    expected: FAIL
+
+  [transport's dtlsCipher]
+    expected: FAIL
+
+  [transport's dtlsRole]
+    expected: FAIL
+
+  [transport's srtpCipher]
+    expected: FAIL
+
+  [transport's selectedCandidatePairChanges]
+    expected: FAIL
+
+  [transport's timestamp]
+    expected: FAIL
+
+  [transport's type]
+    expected: FAIL
+
+  [transport's id]
+    expected: FAIL
+
+  [candidate-pair's packetsSent]
+    expected: FAIL
+
+  [candidate-pair's packetsReceived]
+    expected: FAIL
+
+  [candidate-pair's totalRoundTripTime]
+    expected: FAIL
+
+  [candidate-pair's currentRoundTripTime]
+    expected: FAIL
+
+  [candidate-pair's availableOutgoingBitrate]
+    expected: FAIL
+
+  [candidate-pair's availableIncomingBitrate]
+    expected: PRECONDITION_FAILED
+
+  [candidate-pair's requestsReceived]
+    expected: FAIL
+
+  [candidate-pair's requestsSent]
+    expected: FAIL
+
+  [candidate-pair's responsesReceived]
+    expected: FAIL
+
+  [candidate-pair's responsesSent]
+    expected: FAIL
+
+  [candidate-pair's consentRequestsSent]
+    expected: FAIL
+
+  [candidate-pair's packetsDiscardedOnSend]
+    expected: FAIL
+
+  [candidate-pair's bytesDiscardedOnSend]
+    expected: FAIL
+
+  [local-candidate's transportId]
+    expected: FAIL
+
+  [local-candidate's url]
+    expected: PRECONDITION_FAILED
+
+  [local-candidate's relayProtocol]
+    expected: PRECONDITION_FAILED
+
+  [local-candidate's foundation]
+    expected: FAIL
+
+  [local-candidate's relatedAddress]
+    expected: PRECONDITION_FAILED
+
+  [local-candidate's relatedPort]
+    expected: PRECONDITION_FAILED
+
+  [local-candidate's usernameFragment]
+    expected: FAIL
+
+  [local-candidate's tcpType]
+    expected: FAIL
+
+  [remote-candidate's transportId]
+    expected: FAIL
+
+  [remote-candidate's url]
+    expected: PRECONDITION_FAILED
+
+  [remote-candidate's relayProtocol]
+    expected: PRECONDITION_FAILED
+
+  [remote-candidate's foundation]
+    expected: FAIL
+
+  [remote-candidate's relatedAddress]
+    expected: PRECONDITION_FAILED
+
+  [remote-candidate's relatedPort]
+    expected: PRECONDITION_FAILED
+
+  [remote-candidate's usernameFragment]
+    expected: FAIL
+
+  [remote-candidate's tcpType]
+    expected: PRECONDITION_FAILED
+
+  [certificate's fingerprint]
+    expected: FAIL
+
+  [certificate's fingerprintAlgorithm]
+    expected: FAIL
+
+  [certificate's base64Certificate]
+    expected: FAIL
+
+  [certificate's issuerCertificateId]
+    expected: PRECONDITION_FAILED
+
+  [certificate's timestamp]
+    expected: FAIL
+
+  [certificate's type]
+    expected: FAIL
+
+  [certificate's id]
+    expected: FAIL
+
+  [inbound-rtp's framesRendered]
+    expected: FAIL
+
+  [outbound-rtp's scalabilityMode]
+    expected: FAIL
+
+  [media-playout's kind]
+    expected: FAIL
+
+  [inbound-rtp's retransmittedPacketsReceived]
+    expected: FAIL
+
+  [inbound-rtp's retransmittedBytesReceived]
+    expected: FAIL
+
+  [getStats succeeds]
+    expected:
+      if (os == "android") and not debug: [PASS, TIMEOUT]
+
+  [data-channel's label]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's dataChannelIdentifier]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's id]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's messagesSent]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's bytesReceived]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's state]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's protocol]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's timestamp]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's messagesReceived]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's type]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [data-channel's bytesSent]
+    expected:
+      if (os == "android") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's fecBytesReceived]
+    expected: FAIL
+
+  [inbound-rtp's rtxSsrc]
+    expected: FAIL
+
+  [inbound-rtp's fecSsrc]
+    expected: PRECONDITION_FAILED
+
+  [outbound-rtp's rtxSsrc]
+    expected: FAIL
+
+  [outbound-rtp's qpSum]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's totalInterFrameDelay]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's nackCount]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's framesDecoded]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's totalSquaredInterFrameDelay]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's framesDropped]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's pliCount]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's totalProcessingDelay]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's framesReceived]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's totalDecodeTime]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's frameHeight]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's framesPerSecond]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's firCount]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]
+
+  [inbound-rtp's frameWidth]
+    expected:
+      if (processor == "x86") and not debug: [PASS, FAIL]