BASELINE: Update Chromium to 85.0.4183.14085-based

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

1 files changed, 52 insertions, 0 deletions

diff --git a/chromium/third_party/webrtc/webrtc.gni b/chromium/third_party/webrtc/webrtc.gni
index 4f1d0017f8e..680762f3a14 100644
--- a/chromium/third_party/webrtc/webrtc.gni
+++ b/chromium/third_party/webrtc/webrtc.gni
@@ -155,6 +155,9 @@ declare_args() {
   rtc_use_h264 =
       proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
 
+  # Enable this flag to make webrtc::Mutex be implemented by absl::Mutex.
+  rtc_use_absl_mutex = false
+
   # By default, use normal platform audio support or dummy audio, but don't
   # use file-based audio playout and record.
   rtc_use_dummy_audio_file_devices = false
@@ -323,16 +326,19 @@ set_defaults("rtc_test") {
 set_defaults("rtc_library") {
   configs = rtc_add_configs
   suppressed_configs = []
+  absl_deps = []
 }
 
 set_defaults("rtc_source_set") {
   configs = rtc_add_configs
   suppressed_configs = []
+  absl_deps = []
 }
 
 set_defaults("rtc_static_library") {
   configs = rtc_add_configs
   suppressed_configs = []
+  absl_deps = []
 }
 
 set_defaults("rtc_executable") {
@@ -525,6 +531,20 @@ template("rtc_source_set") {
     if (defined(invoker.public_configs)) {
       public_configs += invoker.public_configs
     }
+
+    # If absl_deps is [], no action is needed. If not [], then it needs to be
+    # converted to //third_party/abseil-cpp:absl when build_with_chromium=true
+    # otherwise it just needs to be added to deps.
+    if (absl_deps != []) {
+      if (!defined(deps)) {
+        deps = []
+      }
+      if (build_with_chromium) {
+        deps += [ "//third_party/abseil-cpp:absl" ]
+      } else {
+        deps += absl_deps
+      }
+    }
   }
 }
 
@@ -600,6 +620,20 @@ template("rtc_static_library") {
     if (defined(invoker.public_configs)) {
       public_configs += invoker.public_configs
     }
+
+    # If absl_deps is [], no action is needed. If not [], then it needs to be
+    # converted to //third_party/abseil-cpp:absl when build_with_chromium=true
+    # otherwise it just needs to be added to deps.
+    if (absl_deps != []) {
+      if (!defined(deps)) {
+        deps = []
+      }
+      if (build_with_chromium) {
+        deps += [ "//third_party/abseil-cpp:absl" ]
+      } else {
+        deps += absl_deps
+      }
+    }
   }
 }
 
@@ -712,6 +746,20 @@ template("rtc_library") {
     if (defined(invoker.public_configs)) {
       public_configs += invoker.public_configs
     }
+
+    # If absl_deps is [], no action is needed. If not [], then it needs to be
+    # converted to //third_party/abseil-cpp:absl when build_with_chromium=true
+    # otherwise it just needs to be added to deps.
+    if (absl_deps != []) {
+      if (!defined(deps)) {
+        deps = []
+      }
+      if (build_with_chromium) {
+        deps += [ "//third_party/abseil-cpp:absl" ]
+      } else {
+        deps += absl_deps
+      }
+    }
   }
 }
 
@@ -1002,6 +1050,7 @@ if (is_android) {
       }
 
       no_build_hooks = true
+      not_needed([ "android_manifest" ])
     }
   }
 
@@ -1020,6 +1069,9 @@ if (is_android) {
       errorprone_args = []
       errorprone_args += [ "-Werror" ]
 
+      # Use WebRTC-specific android lint suppressions file.
+      lint_suppressions_file = "//tools_webrtc/android/suppressions.xml"
+
       if (!defined(deps)) {
         deps = []
       }