Update libwebrtc to M115

https://bugs.webkit.org/show_bug.cgi?id=258208
rdar://107371861

Reviewed by Eric Carlson.

Update libwebrtc to M115 branch.
This allows enabling a few tests.
We have to change a bit RTCDataChannel::didChangeReadyState since connecting is now firing for remote non negotiated channels.
Update WebCore call sites to use updated API.
In particular, callbaks are now more in use which simplifies our code a bit.

* LayoutTests/TestExpectations:
* LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt:
* LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpParameters-encodings-expected.txt:
* LayoutTests/imported/w3c/web-platform-tests/webrtc/RollbackEvents.https-expected.txt:
* LayoutTests/imported/w3c/web-platform-tests/webrtc/protocol/msid-generate-expected.txt:
* LayoutTests/imported/w3c/web-platform-tests/webrtc/protocol/msid-parse-expected.txt:
* Source/ThirdParty/libwebrtc/Configurations/libwebrtc.exp:
* Source/ThirdParty/libwebrtc/Source/third_party/pffft: Resynced.
* Source/ThirdParty/libwebrtc/Source/webrtc: Resynced.
* Source/WebCore/Modules/mediastream/RTCDataChannel.cpp:
(WebCore::RTCDataChannel::didChangeReadyState):
* Source/WebCore/Modules/mediastream/RTCStatsReport.h:
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCCertificateGenerator.cpp:
(WebCore::LibWebRTCCertificateGenerator::RTCCertificateGeneratorCallbackWrapper::create):
(WebCore::LibWebRTCCertificateGenerator::RTCCertificateGeneratorCallbackWrapper::process):
(WebCore::LibWebRTCCertificateGenerator::RTCCertificateGeneratorCallbackWrapper::RTCCertificateGeneratorCallbackWrapper):
(WebCore::LibWebRTCCertificateGenerator::generateCertificate):
(WebCore::LibWebRTCCertificateGenerator::RTCCertificateGeneratorCallback::RTCCertificateGeneratorCallback): Deleted.
(WebCore::LibWebRTCCertificateGenerator::RTCCertificateGeneratorCallback::AddRef const): Deleted.
(WebCore::LibWebRTCCertificateGenerator::RTCCertificateGeneratorCallback::Release const): Deleted.
(): Deleted.
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
(WebCore::LibWebRTCMediaEndpoint::addTrack):
(WebCore::LibWebRTCMediaEndpoint::gatherDecoderImplementationName):
(WebCore::LibWebRTCMediaEndpoint::createSourceAndRTCTrack):
(WebCore::LibWebRTCMediaEndpoint::OnStatsDelivered):
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.cpp:
(WebCore::LibWebRTCRtpSenderTransformBackend::requestKeyFrame):
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransformableFrame.cpp:
(WebCore::LibWebRTCRtpTransformableFrame::videoMetadata const):
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp:
(WebCore::fillRTCStats):
(WebCore::fillRtpStreamStats):
(WebCore::fillReceivedRtpStreamStats):
(WebCore::fillInboundRtpStreamStats):
(WebCore::fillSentRtpStreamStats):
(WebCore::fillOutboundRtpStreamStats):
(WebCore::initializeRTCStatsReportBackingMap):
* Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp:
(WebCore::LibWebRTCProvider::callOnWebRTCNetworkThread):
(WebCore::LibWebRTCProvider::callOnWebRTCSignalingThread):
(WebCore::LibWebRTCProvider::createPeerConnection):
(WebCore::ThreadMessageData::ThreadMessageData): Deleted.
(WebCore::PeerConnectionFactoryAndThreads::OnMessage): Deleted.
* Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp:
(WebCore::MockLibWebRTCPeerConnectionFactory::CreateVideoTrack):
* Source/WebCore/testing/MockLibWebRTCPeerConnection.h:
* Source/WebKit/NetworkProcess/webrtc/NetworkRTCProvider.cpp:
(WebKit::NetworkRTCProvider::callOnRTCNetworkThread):
(WebKit::NetworkMessageData::NetworkMessageData): Deleted.
(WebKit::NetworkRTCProvider::OnMessage): Deleted.
* Source/WebKit/NetworkProcess/webrtc/NetworkRTCProvider.h:
* Source/WebKit/WebProcess/Network/webrtc/LibWebRTCResolver.cpp:
(WebKit::LibWebRTCResolver::Start):
* Source/WebKit/WebProcess/Network/webrtc/LibWebRTCResolver.h:

Canonical link: https://commits.webkit.org/265320@main

Author: youennf

LayoutTests/imported/w3c/web-platform-tests/media-capabilities/decodingInfo.webrtc-expected.txt

@@ -11,7 +11,6 @@ PASS Test that decodingInfo returns a valid MediaCapabilitiesInfo objects
 PASS Test that decodingInfo returns supported, smooth, and powerEfficient set to false for non-webrtc video content type.
 PASS Test that decodingInfo returns supported, smooth, and powerEfficient set to false for non-webrtc audio content type.
 PASS Test that decodingInfo returns supported true for the codec audio/opus returned by RTCRtpReceiver.getCapabilities()
-PASS Test that decodingInfo returns supported true for the codec audio/ISAC returned by RTCRtpReceiver.getCapabilities()
 PASS Test that decodingInfo returns supported true for the codec audio/G722 returned by RTCRtpReceiver.getCapabilities()
 PASS Test that decodingInfo returns supported true for the codec audio/PCMU returned by RTCRtpReceiver.getCapabilities()
 PASS Test that decodingInfo returns supported true for the codec audio/PCMA returned by RTCRtpReceiver.getCapabilities()

LayoutTests/imported/w3c/web-platform-tests/media-capabilities/encodingInfo.webrtc-expected.txt

@@ -11,7 +11,6 @@ PASS Test that encodingInfo returns a valid MediaCapabilitiesInfo objects
 PASS Test that encodingInfo returns supported, smooth, and powerEfficient set to false for non-webrtc video content type.
 PASS Test that encodingInfo returns supported, smooth, and powerEfficient set to false for non-webrtc audio content type.
 PASS Test that encodingInfo returns supported true for the codec audio/opus returned by RTCRtpSender.getCapabilities()
-PASS Test that encodingInfo returns supported true for the codec audio/ISAC returned by RTCRtpSender.getCapabilities()
 PASS Test that encodingInfo returns supported true for the codec audio/G722 returned by RTCRtpSender.getCapabilities()
 PASS Test that encodingInfo returns supported true for the codec audio/PCMU returned by RTCRtpSender.getCapabilities()
 PASS Test that encodingInfo returns supported true for the codec audio/PCMA returned by RTCRtpSender.getCapabilities()

LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-capture-video.https-expected.txt

@@ -0,0 +1,5 @@
+
+Harness Error (TIMEOUT), message = null
+
+TIMEOUT Capturing a video element and sending it via PeerConnection Test timed out
+

LayoutTests/imported/w3c/web-platform-tests/webrtc/simulcast/rid-manipulation-expected.txt

@@ -0,0 +1,3 @@
+
+PASS Remote reanswer altering rids does not throw an exception.
+

LayoutTests/imported/w3c/web-platform-tests/webrtc/simulcast/setParameters-encodings.https-expected.txt

@@ -0,0 +1,17 @@
+
+FAIL sRD(simulcast answer) can narrow the simulcast envelope when interrupted by a setParameters promise_test: Unhandled rejection with value: object "InvalidStateError: Failed to set parameters since getParameters() has never been called on this sender"
+FAIL sRD(simulcast offer) can narrow the simulcast envelope when interrupted by a setParameters promise_test: Unhandled rejection with value: object "InvalidStateError: Failed to set parameters since getParameters() has never been called on this sender"
+PASS a simulcast setParameters followed by a sRD(unicast answer) results in keeping the first encoding
+FAIL sRD(unicast answer) interrupted by setParameters(simulcast) results in keeping the first encoding promise_test: Unhandled rejection with value: object "InvalidStateError: Failed to set parameters since getParameters() has never been called on this sender"
+FAIL sRD(unicast reoffer) interrupted by setParameters(simulcast) results in keeping the first encoding promise_test: Unhandled rejection with value: object "InvalidStateError: Failed to set parameters since getParameters() has never been called on this sender"
+FAIL sRD(simulcast answer) interrupted by a setParameters does not result in losing modifications from the setParameters to the encodings that remain promise_test: Unhandled rejection with value: object "InvalidStateError: Failed to set parameters since getParameters() has never been called on this sender"
+FAIL addTrack, then a unicast setParameters, then sRD(simulcast offer) results in simulcast without the settings from setParameters assert_array_equals: lengths differ, expected array ["foo", "bar"] length 2, got [""] length 1
+FAIL addTrack, then sRD(simulcast offer) interrupted by a unicast setParameters results in simulcast without the settings from setParameters assert_array_equals: lengths differ, expected array ["foo", "bar"] length 2, got [""] length 1
+PASS getParameters, then sLD(unicast answer) interrupted by a simulcast setParameters results in unicast
+FAIL Success task for setLocalDescription(answer) clears [[LastReturnedParameters]] assert_unreached: Should have rejected: undefined Reached unreachable code
+FAIL Success task for setRemoteDescription(offer) clears [[LastReturnedParameters]] assert_unreached: Should have rejected: undefined Reached unreachable code
+FAIL Success task for setRemoteDescription(answer) clears [[LastReturnedParameters]] assert_unreached: Should have rejected: undefined Reached unreachable code
+FAIL addTrack, then rollback of sRD(simulcast offer), brings us back to having a single encoding without any previously set parameters assert_array_equals: lengths differ, expected array ["foo", "bar"] length 2, got [""] length 1
+FAIL rollback of a remote offer that disabled a previously negotiated simulcast should restore simulcast along with any previously set parameters assert_array_equals: lengths differ, expected array ["foo", "bar"] length 2, got ["foo"] length 1
+FAIL rollback of sRD(simulcast offer) interrupted by setParameters(simulcast) brings us back to having a single encoding without any previously set parameters assert_array_equals: lengths differ, expected array ["foo", "bar"] length 2, got [""] length 1
+

Source/ThirdParty/libwebrtc/Source/third_party/pffft/BUILD.gn

@@ -1,4 +1,4 @@
-# Copyright 2019 The Chromium Authors. All rights reserved.
+# Copyright 2019 The Chromium Authors
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
@@ -7,6 +7,8 @@ import("//testing/libfuzzer/fuzzer_test.gni")
 import("//testing/test.gni")
 
 config("common_config") {
+  cflags = [ "-Wno-shadow" ]
+
   if (is_win) {
     defines = [
       # Required to use math constants from math.h.
@@ -17,7 +19,8 @@ config("common_config") {
   # PFFFT doesn't support SIMD on some cpus, so build a scalar version.
   if ((current_cpu == "arm" && !arm_use_neon) || current_cpu == "mipsel" ||
       current_cpu == "mips64el" || current_cpu == "ppc64" ||
-      current_cpu == "s390x") {
+      current_cpu == "riscv64" || current_cpu == "s390x" ||
+      current_cpu == "loong64") {
     defines = [ "PFFFT_SIMD_DISABLE" ]
   }
 }

Source/ThirdParty/libwebrtc/Source/third_party/pffft/OWNERS

@@ -1,2 +1,3 @@
-maxmorin@chromium.org
-olka@chromium.org
+alessiob@chromium.org
+peah@chromium.org
+saza@chromium.org

Source/ThirdParty/libwebrtc/Source/third_party/pffft/generate_seed_corpus.py

@@ -1,5 +1,5 @@
 #!/usr/bin/env python
-# Copyright 2019 The Chromium Authors. All rights reserved.
+# Copyright 2019 The Chromium Authors
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 

Source/ThirdParty/libwebrtc/Source/third_party/pffft/pffft_fuzzer.cc

@@ -1,4 +1,4 @@
-// Copyright 2019 The Chromium Authors. All rights reserved.
+// Copyright 2019 The Chromium Authors
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 

Source/ThirdParty/libwebrtc/Source/third_party/pffft/pffft_unittest.cc

@@ -1,4 +1,4 @@
-// Copyright 2019 The Chromium Authors. All rights reserved.
+// Copyright 2019 The Chromium Authors
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.