BASELINE: Update Chromium to 85.0.4183.14085-based

Change-Id: Iaa42f4680837c57725b1344f108c0196741f6057
Reviewed-by: Allan Sandfeld Jensen <allan.jensen@qt.io>

8 files changed, 14 insertions, 95 deletions

diff --git a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.h b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.h
index 4e9c674ef8e..86eaa6cee5d 100644
--- a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.h
+++ b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.h
@@ -198,18 +198,6 @@ RTC_OBJC_EXPORT
 @property(nonatomic, assign) BOOL allowCodecSwitching;
 
 /**
- * If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
- * that it should use the MediaTransportInterface.
- */
-@property(nonatomic, assign) BOOL useMediaTransport;
-
-/**
- * If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
- * that it should use the MediaTransportInterface for data channels.
- */
-@property(nonatomic, assign) BOOL useMediaTransportForDataChannels;
-
-/**
  * Defines advanced optional cryptographic settings related to SRTP and
  * frame encryption for native WebRTC. Setting this will overwrite any
  * options set through the PeerConnectionFactory (which is deprecated).
diff --git a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm
index 52c14505054..55abbcdb184 100644
--- a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm
+++ b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -52,8 +52,6 @@
 @synthesize turnCustomizer = _turnCustomizer;
 @synthesize activeResetSrtpParams = _activeResetSrtpParams;
 @synthesize allowCodecSwitching = _allowCodecSwitching;
-@synthesize useMediaTransport = _useMediaTransport;
-@synthesize useMediaTransportForDataChannels = _useMediaTransportForDataChannels;
 @synthesize cryptoOptions = _cryptoOptions;
 @synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
 @synthesize rtcpVideoReportIntervalMs = _rtcpVideoReportIntervalMs;
@@ -106,8 +104,6 @@
     _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
     _iceBackupCandidatePairPingInterval =
         config.ice_backup_candidate_pair_ping_interval;
-    _useMediaTransport = config.use_media_transport;
-    _useMediaTransportForDataChannels = config.use_media_transport_for_data_channels;
     _keyType = RTCEncryptionKeyTypeECDSA;
     _iceCandidatePoolSize = config.ice_candidate_pool_size;
     _shouldPruneTurnPorts = config.prune_turn_ports;
@@ -143,7 +139,7 @@
 - (NSString *)description {
   static NSString *formatString = @"RTC_OBJC_TYPE(RTCConfiguration): "
                                   @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n"
-                                  @"%d\n%@\n%d\n%d\n%d\n%d\n%d\n%@\n%d\n}\n";
+                                  @"%d\n%@\n%d\n%d\n%d\n%d\n%d\n%@\n}\n";
 
   return [NSString
       stringWithFormat:formatString,
@@ -169,7 +165,6 @@
                        _disableIPV6OnWiFi,
                        _maxIPv6Networks,
                        _activeResetSrtpParams,
-                       _useMediaTransport,
                        _enableDscp];
 }
 
@@ -208,8 +203,6 @@
       _iceConnectionReceivingTimeout;
   nativeConfig->ice_backup_candidate_pair_ping_interval =
       _iceBackupCandidatePairPingInterval;
-  nativeConfig->use_media_transport = _useMediaTransport;
-  nativeConfig->use_media_transport_for_data_channels = _useMediaTransportForDataChannels;
   rtc::KeyType keyType =
       [[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
   if (_certificate != nullptr) {
diff --git a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm
index fa68d08e74d..9e561fc65f9 100644
--- a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm
+++ b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm
@@ -29,7 +29,6 @@
 
 #include "api/jsep_ice_candidate.h"
 #include "api/rtc_event_log_output_file.h"
-#include "api/transport/media/media_transport_interface.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 
diff --git a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
index c2aab0be568..1d3b82550a5 100644
--- a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
+++ b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
@@ -17,7 +17,6 @@ namespace webrtc {
 class AudioDeviceModule;
 class AudioEncoderFactory;
 class AudioDecoderFactory;
-class MediaTransportFactory;
 class NetworkControllerFactoryInterface;
 class VideoEncoderFactory;
 class VideoDecoderFactory;
@@ -65,30 +64,12 @@ NS_ASSUME_NONNULL_BEGIN
                     audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule
                 audioProcessingModule:
                     (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
-                mediaTransportFactory:
-                    (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
-
-- (instancetype)
-    initWithNativeAudioEncoderFactory:
-        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
-            nativeAudioDecoderFactory:
-                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
-            nativeVideoEncoderFactory:
-                (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
-            nativeVideoDecoderFactory:
-                (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
-                    audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule
-                audioProcessingModule:
-                    (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
              networkControllerFactory:(std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
-                                          networkControllerFactory
-                mediaTransportFactory:
-                    (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
+                                          networkControllerFactory;
 
 - (instancetype)
     initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
-            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
-     mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
 
 /** Initialize an RTCPeerConnection with a configuration, constraints, and
  *  dependencies.
diff --git a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
index 2e34b05fed0..4ce38dbd7fd 100644
--- a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
+++ b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
@@ -52,7 +52,6 @@
 // C++ target.
 // TODO(zhihuang): Remove nogncheck once MediaEngineInterface is moved to C++
 // API layer.
-#include "api/transport/media/media_transport_interface.h"
 #include "media/engine/webrtc_media_engine.h"  // nogncheck
 
 @implementation RTC_OBJC_TYPE (RTCPeerConnectionFactory) {
@@ -84,15 +83,13 @@
               nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory([[RTC_OBJC_TYPE(
                                             RTCVideoDecoderFactoryH264) alloc] init])
                       audioDeviceModule:[self audioDeviceModule]
-                  audioProcessingModule:nullptr
-                  mediaTransportFactory:nullptr];
+                  audioProcessingModule:nullptr];
 #endif
 }
 
 - (instancetype)
     initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
-            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
-     mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory {
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory {
 #ifdef HAVE_NO_MEDIA
   return [self initWithNoMedia];
 #else
@@ -109,18 +106,9 @@
                        nativeVideoEncoderFactory:std::move(native_encoder_factory)
                        nativeVideoDecoderFactory:std::move(native_decoder_factory)
                                audioDeviceModule:[self audioDeviceModule]
-                           audioProcessingModule:nullptr
-                           mediaTransportFactory:std::move(mediaTransportFactory)];
+                           audioProcessingModule:nullptr];
 #endif
 }
-- (instancetype)
-    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
-            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory {
-  return [self initWithEncoderFactory:encoderFactory
-                       decoderFactory:decoderFactory
-                mediaTransportFactory:nullptr];
-}
-
 - (instancetype)initNative {
   if (self = [super init]) {
     _networkThread = rtc::Thread::CreateWithSocketServer();
@@ -170,30 +158,7 @@
                        nativeVideoDecoderFactory:std::move(videoDecoderFactory)
                                audioDeviceModule:audioDeviceModule
                            audioProcessingModule:audioProcessingModule
-                           mediaTransportFactory:nullptr];
-}
-
-- (instancetype)initWithNativeAudioEncoderFactory:
-                    (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
-                        nativeAudioDecoderFactory:
-                            (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
-                        nativeVideoEncoderFactory:
-                            (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
-                        nativeVideoDecoderFactory:
-                            (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
-                                audioDeviceModule:(webrtc::AudioDeviceModule *)audioDeviceModule
-                            audioProcessingModule:
-                                (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
-                            mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)
-                                                      mediaTransportFactory {
-  return [self initWithNativeAudioEncoderFactory:audioEncoderFactory
-                       nativeAudioDecoderFactory:audioDecoderFactory
-                       nativeVideoEncoderFactory:std::move(videoEncoderFactory)
-                       nativeVideoDecoderFactory:std::move(videoDecoderFactory)
-                               audioDeviceModule:audioDeviceModule
-                           audioProcessingModule:audioProcessingModule
-                        networkControllerFactory:nullptr
-                           mediaTransportFactory:std::move(mediaTransportFactory)];
+                        networkControllerFactory:nullptr];
 }
 - (instancetype)initWithNativeAudioEncoderFactory:
                     (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
@@ -208,9 +173,7 @@
                                 (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
                          networkControllerFactory:
                              (std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
-                                 networkControllerFactory
-                            mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)
-                                                      mediaTransportFactory {
+                                 networkControllerFactory {
   if (self = [self initNative]) {
     webrtc::PeerConnectionFactoryDependencies dependencies;
     dependencies.network_thread = _networkThread.get();
@@ -235,7 +198,6 @@
     dependencies.event_log_factory =
         std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
     dependencies.network_controller_factory = std::move(networkControllerFactory);
-    dependencies.media_transport_factory = std::move(mediaTransportFactory);
 #endif
     _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
     NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
diff --git a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
index 8f52bea8e33..991ec5a41cc 100644
--- a/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
+++ b/chromium/third_party/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
@@ -13,7 +13,6 @@
 
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
-#include "api/transport/media/media_transport_interface.h"
 #include "api/video_codecs/video_decoder_factory.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "modules/audio_device/include/audio_device.h"
@@ -26,7 +25,6 @@
   rtc::scoped_refptr<webrtc::AudioDecoderFactory> _audioDecoderFactory;
   rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
   rtc::scoped_refptr<webrtc::AudioProcessing> _audioProcessingModule;
-  std::unique_ptr<webrtc::MediaTransportFactory> _mediaTransportFactory;
 }
 
 + (RTCPeerConnectionFactoryBuilder *)builder {
@@ -41,8 +39,7 @@
                           nativeVideoEncoderFactory:std::move(_videoEncoderFactory)
                           nativeVideoDecoderFactory:std::move(_videoDecoderFactory)
                                   audioDeviceModule:_audioDeviceModule
-                              audioProcessingModule:_audioProcessingModule
-                              mediaTransportFactory:std::move(_mediaTransportFactory)];
+                              audioProcessingModule:_audioProcessingModule];
 }
 
 - (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory {
diff --git a/chromium/third_party/webrtc/sdk/objc/native/src/audio/audio_device_ios.mm b/chromium/third_party/webrtc/sdk/objc/native/src/audio/audio_device_ios.mm
index b70c4d0e50b..55dc517e4c1 100644
--- a/chromium/third_party/webrtc/sdk/objc/native/src/audio/audio_device_ios.mm
+++ b/chromium/third_party/webrtc/sdk/objc/native/src/audio/audio_device_ios.mm
@@ -102,7 +102,8 @@ static void LogDeviceInfo() {
 #endif  // !defined(NDEBUG)
 
 AudioDeviceIOS::AudioDeviceIOS()
-    : audio_device_buffer_(nullptr),
+    : MessageHandler(false),
+      audio_device_buffer_(nullptr),
       audio_unit_(nullptr),
       recording_(0),
       playing_(0),
@@ -125,6 +126,7 @@ AudioDeviceIOS::AudioDeviceIOS()
 AudioDeviceIOS::~AudioDeviceIOS() {
   RTC_DCHECK(thread_checker_.IsCurrent());
   LOGI() << "~dtor" << ios::GetCurrentThreadDescription();
+  thread_->Clear(this);
   Terminate();
   audio_session_observer_ = nil;
 }
diff --git a/chromium/third_party/webrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm b/chromium/third_party/webrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
index 7d19d4095d7..14131dc38d3 100644
--- a/chromium/third_party/webrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
+++ b/chromium/third_party/webrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
@@ -22,7 +22,6 @@ extern "C" {
 
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/transport/media/media_transport_interface.h"
 #include "api/video_codecs/video_decoder_factory.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "modules/audio_device/include/audio_device.h"
@@ -50,8 +49,7 @@ extern "C" {
               nativeVideoEncoderFactory:nullptr
               nativeVideoDecoderFactory:nullptr
                       audioDeviceModule:nullptr
-                  audioProcessingModule:nullptr
-                  mediaTransportFactory:nullptr]);
+                  audioProcessingModule:nullptr]);
 #endif
   RTCPeerConnectionFactoryBuilder* builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
   RTC_OBJC_TYPE(RTCPeerConnectionFactory)* peerConnectionFactory =
@@ -72,8 +70,7 @@ extern "C" {
               nativeVideoEncoderFactory:nullptr
               nativeVideoDecoderFactory:nullptr
                       audioDeviceModule:nullptr
-                  audioProcessingModule:nullptr
-                  mediaTransportFactory:nullptr]);
+                  audioProcessingModule:nullptr]);
 #endif
   RTCPeerConnectionFactoryBuilder* builder = [RTCPeerConnectionFactoryBuilder defaultBuilder];
   RTC_OBJC_TYPE(RTCPeerConnectionFactory)* peerConnectionFactory =