BASELINE: Update Chromium to 85.0.4183.14085-based

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

18 files changed, 99 insertions, 191 deletions

diff --git a/chromium/third_party/webrtc/sdk/BUILD.gn b/chromium/third_party/webrtc/sdk/BUILD.gn
index 1b313b3d872..dc9b265155e 100644
--- a/chromium/third_party/webrtc/sdk/BUILD.gn
+++ b/chromium/third_party/webrtc/sdk/BUILD.gn
@@ -35,8 +35,8 @@ rtc_library("media_constraints") {
   deps = [
     "../api:audio_options_api",
     "../api:libjingle_peerconnection_api",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("sdk_tests") {
@@ -272,8 +272,8 @@ if (is_ios || is_mac) {
           "../rtc_base:checks",
           "../system_wrappers:field_trial",
           "../system_wrappers:metrics",
-          "//third_party/abseil-cpp/absl/base:core_headers",
         ]
+        absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
 
         libs = [ "AudioToolbox.framework" ]
       }
@@ -444,8 +444,8 @@ if (is_ios || is_mac) {
         "../media:rtc_media_base",
         "../rtc_base",
         "../rtc_base:checks",
-        "//third_party/abseil-cpp/absl/types:optional",
       ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 
       configs += [
         "..:common_objc",
@@ -946,7 +946,6 @@ if (is_ios || is_mac) {
         "../api/crypto:frame_encryptor_interface",
         "../api/rtc_event_log:rtc_event_log_factory",
         "../api/task_queue:default_task_queue_factory",
-        "../api/transport/media:media_transport_interface",
         "../api/video:video_frame",
         "../api/video:video_rtp_headers",
         "../api/video_codecs:video_codecs_api",
@@ -1217,7 +1216,6 @@ if (is_ios || is_mac) {
           "../api/audio_codecs:audio_codecs_api",
           "../api/audio_codecs:builtin_audio_decoder_factory",
           "../api/audio_codecs:builtin_audio_encoder_factory",
-          "../api/transport/media:media_transport_interface",
           "../api/video_codecs:video_codecs_api",
           "../media:rtc_media_base",
           "../modules:module_api",
@@ -1572,8 +1570,8 @@ if (is_ios || is_mac) {
         "../api/video_codecs:video_codecs_api",
         "../common_video",
         "../rtc_base",
-        "//third_party/abseil-cpp/absl/memory",
       ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
     }
 
     rtc_library("native_video") {
diff --git a/chromium/third_party/webrtc/sdk/android/BUILD.gn b/chromium/third_party/webrtc/sdk/android/BUILD.gn
index f4d05c1fde3..3c824dbe756 100644
--- a/chromium/third_party/webrtc/sdk/android/BUILD.gn
+++ b/chromium/third_party/webrtc/sdk/android/BUILD.gn
@@ -272,7 +272,6 @@ if (is_android) {
       "api/org/webrtc/MediaSource.java",
       "api/org/webrtc/MediaStream.java",
       "api/org/webrtc/MediaStreamTrack.java",
-      "api/org/webrtc/MediaTransportFactoryFactory.java",
       "api/org/webrtc/NativeLibraryLoader.java",
       "api/org/webrtc/NativePeerConnectionFactory.java",
       "api/org/webrtc/NetEqFactoryFactory.java",
@@ -326,7 +325,7 @@ if (is_android) {
       "//third_party/android_deps:com_android_support_support_annotations_java",
     ]
     srcjar_deps = [
-      "//api:rtp_parameters_enums",
+      "//api:priority_enums",
       "//api/video:video_frame_enums",
     ]
   }
@@ -558,8 +557,8 @@ if (current_os == "linux" || is_android) {
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:field_trial",
       "../../system_wrappers:metrics",
-      "//third_party/abseil-cpp/absl/types:optional",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
   }
 
   rtc_library("audio_jni") {
@@ -655,9 +654,9 @@ if (current_os == "linux" || is_android) {
       "../../rtc_base:checks",
       "../../rtc_base:rtc_task_queue",
       "../../rtc_base/task_utils:to_queued_task",
-      "//third_party/abseil-cpp/absl/types:optional",
       "//third_party/libyuv",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
   }
 
   rtc_library("peerconnection_jni") {
@@ -745,6 +744,8 @@ if (current_os == "linux" || is_android) {
       "../../rtc_base:rtc_task_queue",
       "../../rtc_base/system:thread_registry",
       "../../system_wrappers:field_trial",
+    ]
+    absl_deps = [
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/types:optional",
     ]
@@ -837,8 +838,8 @@ if (current_os == "linux" || is_android) {
       "//api:array_view",
       "//rtc_base:checks",
       "//rtc_base:rtc_base_approved",
-      "//third_party/abseil-cpp/absl/types:optional",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
   }
 
   rtc_library("native_api_base") {
@@ -932,8 +933,8 @@ if (current_os == "linux" || is_android) {
       "../../rtc_base:criticalsection",
       "../../rtc_base:logging",
       "../../rtc_base:stringutils",
-      "//third_party/abseil-cpp/absl/base:core_headers",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
   }
 
   # API for creating C++ wrapper implementations of api/mediastreaminterface.h
@@ -1042,8 +1043,8 @@ if (current_os == "linux" || is_android) {
       "../../rtc_base:checks",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:metrics",
-      "//third_party/abseil-cpp/absl/types:optional",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
   }
 
   rtc_library("java_audio_device_module") {
@@ -1065,8 +1066,8 @@ if (current_os == "linux" || is_android) {
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:field_trial",
       "../../system_wrappers:metrics",
-      "//third_party/abseil-cpp/absl/types:optional",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
   }
 
   if (rtc_enable_android_aaudio) {
@@ -1092,8 +1093,8 @@ if (current_os == "linux" || is_android) {
         "../../rtc_base:checks",
         "../../rtc_base:rtc_base_approved",
         "../../system_wrappers",
-        "//third_party/abseil-cpp/absl/types:optional",
       ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
     }
   }
 
@@ -1118,8 +1119,8 @@ if (current_os == "linux" || is_android) {
       "../../modules/audio_device:audio_device_buffer",
       "../../rtc_base:checks",
       "../../rtc_base:rtc_base_approved",
-      "//third_party/abseil-cpp/absl/types:optional",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
   }
 
   #########################
@@ -1344,9 +1345,16 @@ if (is_android) {
 
       deps = [
         ":base_java",
+        ":builtin_audio_codecs_java",
+        ":camera_java",
         ":default_video_codec_factory_java",
+        ":filevideo_java",
+        ":hwcodecs_java",
         ":libjingle_peerconnection_java",
         ":libjingle_peerconnection_metrics_default_java",
+        ":peerconnection_java",
+        ":surfaceviewrenderer_java",
+        ":swcodecs_java",
         ":video_api_java",
         ":video_java",
         "//base:base_java_test_support",
@@ -1354,6 +1362,7 @@ if (is_android) {
         "//third_party/android_support_test_runner:rules_java",
         "//third_party/android_support_test_runner:runner_java",
         "//third_party/google-truth:google_truth_java",
+        "//third_party/hamcrest:hamcrest_library_java",
         "//third_party/junit",
         "//third_party/mockito:mockito_java",
       ]
@@ -1450,8 +1459,8 @@ if (is_android) {
       "../../test:fileutils",
       "../../test:test_support",
       "../../testing/gtest",
-      "//third_party/abseil-cpp/absl/memory",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
   }
 
   rtc_android_library("native_unittests_java") {
@@ -1508,7 +1517,13 @@ if (is_android) {
     ]
 
     deps = [
+      ":base_java",
+      ":camera_java",
+      ":hwcodecs_java",
       ":libjingle_peerconnection_java",
+      ":peerconnection_java",
+      ":video_api_java",
+      ":video_java",
       "//base:base_java_test_support",
       "//third_party/google-truth:google_truth_java",
     ]
diff --git a/chromium/third_party/webrtc/sdk/android/api/org/webrtc/MediaTransportFactoryFactory.java b/chromium/third_party/webrtc/sdk/android/api/org/webrtc/MediaTransportFactoryFactory.java
deleted file mode 100644
index c16a37a6d77..00000000000
--- a/chromium/third_party/webrtc/sdk/android/api/org/webrtc/MediaTransportFactoryFactory.java
+++ /dev/null
@@ -1,22 +0,0 @@
-/*
- *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc;
-
-/**
- * Factory for creating webrtc::MediaTransportFactory instances.
- */
-public interface MediaTransportFactoryFactory {
-  /**
-   * Dynamically allocates a webrtc::MediaTransportFactory instance and returns a pointer to it.
-   * The caller takes ownership of the object.
-   */
-  public long createNativeMediaTransportFactory();
-}
diff --git a/chromium/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnection.java b/chromium/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnection.java
index bf0a2e94413..920e2f12c96 100644
--- a/chromium/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnection.java
+++ b/chromium/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnection.java
@@ -536,18 +536,6 @@ public class PeerConnection {
     // Null indicates no change to currently configured value.
     @Nullable public Boolean allowCodecSwitching;
 
-    /*
-     * Experimental flag that enables a use of media transport. If this is true, the media transport
-     * factory MUST be provided to the PeerConnectionFactory.
-     */
-    public boolean useMediaTransport;
-
-    /*
-     * Experimental flag that enables a use of media transport for data channels. If this is true,
-     * the media transport factory MUST be provided to the PeerConnectionFactory.
-     */
-    public boolean useMediaTransportForDataChannels;
-
     /**
      * Defines advanced optional cryptographic settings related to SRTP and
      * frame encryption for native WebRTC. Setting this will overwrite any
@@ -602,8 +590,6 @@ public class PeerConnection {
       networkPreference = AdapterType.UNKNOWN;
       sdpSemantics = SdpSemantics.PLAN_B;
       activeResetSrtpParams = false;
-      useMediaTransport = false;
-      useMediaTransportForDataChannels = false;
       cryptoOptions = null;
       turnLoggingId = null;
       allowCodecSwitching = null;
@@ -816,16 +802,6 @@ public class PeerConnection {
       return allowCodecSwitching;
     }
 
-    @CalledByNative("RTCConfiguration")
-    boolean getUseMediaTransport() {
-      return useMediaTransport;
-    }
-
-    @CalledByNative("RTCConfiguration")
-    boolean getUseMediaTransportForDataChannels() {
-      return useMediaTransportForDataChannels;
-    }
-
     @Nullable
     @CalledByNative("RTCConfiguration")
     CryptoOptions getCryptoOptions() {
diff --git a/chromium/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java b/chromium/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
index decdc0cc427..c87e639f235 100644
--- a/chromium/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
+++ b/chromium/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
@@ -175,7 +175,6 @@ public class PeerConnectionFactory {
     @Nullable private FecControllerFactoryFactoryInterface fecControllerFactoryFactory;
     @Nullable private NetworkControllerFactoryFactory networkControllerFactoryFactory;
     @Nullable private NetworkStatePredictorFactoryFactory networkStatePredictorFactoryFactory;
-    @Nullable private MediaTransportFactoryFactory mediaTransportFactoryFactory;
     @Nullable private NetEqFactoryFactory neteqFactoryFactory;
 
     private Builder() {}
@@ -247,13 +246,6 @@ public class PeerConnectionFactory {
       return this;
     }
 
-    /** Sets a MediaTransportFactoryFactory for a PeerConnectionFactory. */
-    public Builder setMediaTransportFactoryFactory(
-        MediaTransportFactoryFactory mediaTransportFactoryFactory) {
-      this.mediaTransportFactoryFactory = mediaTransportFactoryFactory;
-      return this;
-    }
-
     /**
      * Sets a NetEqFactoryFactory for the PeerConnectionFactory. When using a
      * custom NetEqFactoryFactory, the AudioDecoderFactoryFactory will be set
@@ -284,9 +276,6 @@ public class PeerConnectionFactory {
           networkStatePredictorFactoryFactory == null
               ? 0
               : networkStatePredictorFactoryFactory.createNativeNetworkStatePredictorFactory(),
-          mediaTransportFactoryFactory == null
-              ? 0
-              : mediaTransportFactoryFactory.createNativeMediaTransportFactory(),
           neteqFactoryFactory == null ? 0 : neteqFactoryFactory.createNativeNetEqFactory());
     }
   }
@@ -607,7 +596,7 @@ public class PeerConnectionFactory {
       long audioDecoderFactory, VideoEncoderFactory encoderFactory,
       VideoDecoderFactory decoderFactory, long nativeAudioProcessor,
       long nativeFecControllerFactory, long nativeNetworkControllerFactory,
-      long nativeNetworkStatePredictorFactory, long mediaTransportFactory, long neteqFactory);
+      long nativeNetworkStatePredictorFactory, long neteqFactory);
 
   private static native long nativeCreatePeerConnection(long factory,
       PeerConnection.RTCConfiguration rtcConfig, MediaConstraints constraints, long nativeObserver,
diff --git a/chromium/third_party/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java b/chromium/third_party/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
index 3522a874879..b7b028f5aeb 100644
--- a/chromium/third_party/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
+++ b/chromium/third_party/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
@@ -263,6 +263,17 @@ public class SurfaceTextureHelper {
     });
   }
 
+  /**
+   * Forces a frame to be produced. If no new frame is available, the last frame is sent to the
+   * listener again.
+   */
+  public void forceFrame() {
+    handler.post(() -> {
+      hasPendingTexture = true;
+      tryDeliverTextureFrame();
+    });
+  }
+
   /** Set the rotation of the delivered frames. */
   public void setFrameRotation(int rotation) {
     handler.post(() -> this.frameRotation = rotation);
diff --git a/chromium/third_party/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java b/chromium/third_party/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
index edc9dd179df..94eb2a4357a 100644
--- a/chromium/third_party/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
+++ b/chromium/third_party/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
@@ -183,7 +183,7 @@ class WebRtcAudioTrack {
   }
 
   @CalledByNative
-  private boolean initPlayout(int sampleRate, int channels, double bufferSizeFactor) {
+  private int initPlayout(int sampleRate, int channels, double bufferSizeFactor) {
     threadChecker.checkIsOnValidThread();
     Logging.d(TAG,
         "initPlayout(sampleRate=" + sampleRate + ", channels=" + channels
@@ -212,14 +212,14 @@ class WebRtcAudioTrack {
     // can happen that |minBufferSizeInBytes| contains an invalid value.
     if (minBufferSizeInBytes < byteBuffer.capacity()) {
       reportWebRtcAudioTrackInitError("AudioTrack.getMinBufferSize returns an invalid value.");
-      return false;
+      return -1;
     }
 
     // Ensure that prevision audio session was stopped correctly before trying
     // to create a new AudioTrack.
     if (audioTrack != null) {
       reportWebRtcAudioTrackInitError("Conflict with existing AudioTrack.");
-      return false;
+      return -1;
     }
     try {
       // Create an AudioTrack object and initialize its associated audio buffer.
@@ -241,7 +241,7 @@ class WebRtcAudioTrack {
     } catch (IllegalArgumentException e) {
       reportWebRtcAudioTrackInitError(e.getMessage());
       releaseAudioResources();
-      return false;
+      return -1;
     }
 
     // It can happen that an AudioTrack is created but it was not successfully
@@ -250,11 +250,11 @@ class WebRtcAudioTrack {
     if (audioTrack == null || audioTrack.getState() != AudioTrack.STATE_INITIALIZED) {
       reportWebRtcAudioTrackInitError("Initialization of audio track failed.");
       releaseAudioResources();
-      return false;
+      return -1;
     }
     logMainParameters();
     logMainParametersExtended();
-    return true;
+    return minBufferSizeInBytes;
   }
 
   @CalledByNative
@@ -423,6 +423,14 @@ class WebRtcAudioTrack {
     }
   }
 
+  @CalledByNative
+  private int getBufferSizeInFrames() {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) {
+      return audioTrack.getBufferSizeInFrames();
+    }
+    return -1;
+  }
+
   private void logBufferCapacityInFrames() {
     if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.N) {
       Logging.d(TAG,
diff --git a/chromium/third_party/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.cc b/chromium/third_party/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.cc
index 8f0a0417110..d5b880b1b0f 100644
--- a/chromium/third_party/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.cc
+++ b/chromium/third_party/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.cc
@@ -20,6 +20,7 @@
 #include "sdk/android/generated_java_audio_device_module_native_jni/WebRtcAudioTrack_jni.h"
 #include "sdk/android/src/jni/jni_helpers.h"
 #include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
 
 namespace webrtc {
 
@@ -89,12 +90,33 @@ int32_t AudioTrackJni::InitPlayout() {
              nullptr);
   if (buffer_size_factor == 0)
     buffer_size_factor = 1.0;
-  if (!Java_WebRtcAudioTrack_initPlayout(
-          env_, j_audio_track_, audio_parameters_.sample_rate(),
-          static_cast<int>(audio_parameters_.channels()), buffer_size_factor)) {
+  int requested_buffer_size_bytes = Java_WebRtcAudioTrack_initPlayout(
+      env_, j_audio_track_, audio_parameters_.sample_rate(),
+      static_cast<int>(audio_parameters_.channels()), buffer_size_factor);
+  if (requested_buffer_size_bytes < 0) {
     RTC_LOG(LS_ERROR) << "InitPlayout failed";
     return -1;
   }
+  // Update UMA histograms for both the requested and actual buffer size.
+  // To avoid division by zero, we assume the sample rate is 48k if an invalid
+  // value is found.
+  const int sample_rate = audio_parameters_.sample_rate() <= 0
+                              ? 48000
+                              : audio_parameters_.sample_rate();
+  // This calculation assumes that audio is mono.
+  const int requested_buffer_size_ms =
+      (requested_buffer_size_bytes * 1000) / (2 * sample_rate);
+  RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeRequestedAudioBufferSizeMs",
+                       requested_buffer_size_ms, 0, 1000, 100);
+  int actual_buffer_size_frames =
+      Java_WebRtcAudioTrack_getBufferSizeInFrames(env_, j_audio_track_);
+  if (actual_buffer_size_frames >= 0) {
+    const int actual_buffer_size_ms =
+        actual_buffer_size_frames * 1000 / sample_rate;
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeAudioBufferSizeMs",
+                         actual_buffer_size_ms, 0, 1000, 100);
+  }
+
   initialized_ = true;
   return 0;
 }
diff --git a/chromium/third_party/webrtc/sdk/android/src/jni/pc/peer_connection.cc b/chromium/third_party/webrtc/sdk/android/src/jni/pc/peer_connection.cc
index 0ae39fbf661..05e940ee806 100644
--- a/chromium/third_party/webrtc/sdk/android/src/jni/pc/peer_connection.cc
+++ b/chromium/third_party/webrtc/sdk/android/src/jni/pc/peer_connection.cc
@@ -264,11 +264,6 @@ void JavaToNativeRTCConfiguration(
   rtc_config->sdp_semantics = JavaToNativeSdpSemantics(jni, j_sdp_semantics);
   rtc_config->active_reset_srtp_params =
       Java_RTCConfiguration_getActiveResetSrtpParams(jni, j_rtc_config);
-  rtc_config->use_media_transport =
-      Java_RTCConfiguration_getUseMediaTransport(jni, j_rtc_config);
-  rtc_config->use_media_transport_for_data_channels =
-      Java_RTCConfiguration_getUseMediaTransportForDataChannels(jni,
-                                                                j_rtc_config);
   rtc_config->crypto_options =
       JavaToNativeOptionalCryptoOptions(jni, j_crypto_options);
 
diff --git a/chromium/third_party/webrtc/sdk/android/src/jni/pc/peer_connection_factory.cc b/chromium/third_party/webrtc/sdk/android/src/jni/pc/peer_connection_factory.cc
index 48dd6e41d89..32382fe5216 100644
--- a/chromium/third_party/webrtc/sdk/android/src/jni/pc/peer_connection_factory.cc
+++ b/chromium/third_party/webrtc/sdk/android/src/jni/pc/peer_connection_factory.cc
@@ -82,7 +82,9 @@ void PostJavaCallback(JNIEnv* env,
     JavaAsyncCallback(JNIEnv* env,
                       const JavaRef<jobject>& j_object,
                       JavaMethodPointer java_method_pointer)
-        : j_object_(env, j_object), java_method_pointer_(java_method_pointer) {}
+        : rtc::MessageHandler(false),
+          j_object_(env, j_object),
+          java_method_pointer_(java_method_pointer) {}
 
     void OnMessage(rtc::Message*) override {
       java_method_pointer_(AttachCurrentThreadIfNeeded(), j_object_);
@@ -246,7 +248,7 @@ static void JNI_PeerConnectionFactory_ShutdownInternalTracer(JNIEnv* jni) {
 
 // Following parameters are optional:
 // |audio_device_module|, |jencoder_factory|, |jdecoder_factory|,
-// |audio_processor|, |media_transport_factory|, |fec_controller_factory|,
+// |audio_processor|, |fec_controller_factory|,
 // |network_state_predictor_factory|, |neteq_factory|.
 ScopedJavaLocalRef<jobject> CreatePeerConnectionFactoryForJava(
     JNIEnv* jni,
@@ -263,7 +265,6 @@ ScopedJavaLocalRef<jobject> CreatePeerConnectionFactoryForJava(
         network_controller_factory,
     std::unique_ptr<NetworkStatePredictorFactoryInterface>
         network_state_predictor_factory,
-    std::unique_ptr<MediaTransportFactory> media_transport_factory,
     std::unique_ptr<NetEqFactory> neteq_factory) {
   // talk/ assumes pretty widely that the current Thread is ThreadManager'd, but
   // ThreadManager only WrapCurrentThread()s the thread where it is first
@@ -310,7 +311,6 @@ ScopedJavaLocalRef<jobject> CreatePeerConnectionFactoryForJava(
       std::move(network_controller_factory);
   dependencies.network_state_predictor_factory =
       std::move(network_state_predictor_factory);
-  dependencies.media_transport_factory = std::move(media_transport_factory);
   dependencies.neteq_factory = std::move(neteq_factory);
 
   cricket::MediaEngineDependencies media_dependencies;
@@ -355,7 +355,6 @@ JNI_PeerConnectionFactory_CreatePeerConnectionFactory(
     jlong native_fec_controller_factory,
     jlong native_network_controller_factory,
     jlong native_network_state_predictor_factory,
-    jlong native_media_transport_factory,
     jlong native_neteq_factory) {
   rtc::scoped_refptr<AudioProcessing> audio_processor =
       reinterpret_cast<AudioProcessing*>(native_audio_processor);
@@ -372,8 +371,6 @@ JNI_PeerConnectionFactory_CreatePeerConnectionFactory(
           native_network_controller_factory),
       TakeOwnershipOfUniquePtr<NetworkStatePredictorFactoryInterface>(
           native_network_state_predictor_factory),
-      TakeOwnershipOfUniquePtr<MediaTransportFactory>(
-          native_media_transport_factory),
       TakeOwnershipOfUniquePtr<NetEqFactory>(native_neteq_factory));
 }
 
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 =