BASELINE: Update Chromium to 85.0.4183.14085-based

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

11 files changed, 280 insertions, 46 deletions

diff --git a/chromium/third_party/webrtc/api/video/BUILD.gn b/chromium/third_party/webrtc/api/video/BUILD.gn
index 290c2f2abba..ee7e51d4c42 100644
--- a/chromium/third_party/webrtc/api/video/BUILD.gn
+++ b/chromium/third_party/webrtc/api/video/BUILD.gn
@@ -21,7 +21,6 @@ rtc_library("video_rtp_headers") {
     "hdr_metadata.h",
     "video_content_type.cc",
     "video_content_type.h",
-    "video_frame_marking.h",
     "video_rotation.h",
     "video_timing.cc",
     "video_timing.h",
@@ -31,8 +30,8 @@ rtc_library("video_rtp_headers") {
     "..:array_view",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("video_frame") {
@@ -56,8 +55,8 @@ rtc_library("video_frame") {
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 if (is_android) {
@@ -141,8 +140,8 @@ rtc_library("encoded_image") {
     "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("encoded_frame") {
@@ -173,8 +172,8 @@ rtc_library("video_bitrate_allocation") {
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("video_bitrate_allocator") {
@@ -209,8 +208,8 @@ rtc_source_set("video_stream_decoder") {
     "../task_queue",
     "../units:time_delta",
     "../video_codecs:video_codecs_api",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("video_stream_decoder_create") {
@@ -237,7 +236,10 @@ rtc_library("video_adaptation") {
     "video_adaptation_reason.h",
   ]
 
-  deps = [ "../../rtc_base:checks" ]
+  deps = [
+    "../../rtc_base:checks",
+    "../../rtc_base:stringutils",
+  ]
 }
 
 rtc_source_set("video_stream_encoder") {
@@ -256,10 +258,29 @@ rtc_source_set("video_stream_encoder") {
     ":video_codec_constants",
     ":video_frame",
     "..:rtp_parameters",
+    "..:scoped_refptr",
     "../:fec_controller_api",
     "../:rtp_parameters",
+    "../adaptation:resource_adaptation_api",
     "../units:data_rate",
     "../video_codecs:video_codecs_api",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_source_set("video_frame_metadata") {
+  visibility = [ "*" ]
+  sources = [
+    "video_frame_metadata.cc",
+    "video_frame_metadata.h",
+  ]
+  deps = [
+    "..:array_view",
+    "../../modules/rtp_rtcp:rtp_video_header",
+    "../transport/rtp:dependency_descriptor",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -299,16 +320,21 @@ rtc_library("builtin_video_bitrate_allocator_factory") {
     "../../modules/video_coding:video_coding_utility",
     "../../modules/video_coding:webrtc_vp9_helpers",
     "../video_codecs:video_codecs_api",
-    "//third_party/abseil-cpp/absl/base:core_headers",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
 }
 
 if (rtc_include_tests) {
   rtc_library("video_unittests") {
     testonly = true
-    sources = [ "video_stream_decoder_create_unittest.cc" ]
+    sources = [
+      "video_frame_metadata_unittest.cc",
+      "video_stream_decoder_create_unittest.cc",
+    ]
     deps = [
+      ":video_frame_metadata",
       ":video_stream_decoder_create",
+      "../../modules/rtp_rtcp:rtp_video_header",
       "../../test:test_support",
       "../task_queue:default_task_queue_factory",
       "../video_codecs:builtin_video_decoder_factory",
diff --git a/chromium/third_party/webrtc/api/video/OWNERS b/chromium/third_party/webrtc/api/video/OWNERS
index 315f85e7d00..e4a16c360a1 100644
--- a/chromium/third_party/webrtc/api/video/OWNERS
+++ b/chromium/third_party/webrtc/api/video/OWNERS
@@ -1,3 +1,4 @@
+brandtr@webrtc.org
 magjed@webrtc.org
 nisse@webrtc.org
 
diff --git a/chromium/third_party/webrtc/api/video/test/BUILD.gn b/chromium/third_party/webrtc/api/video/test/BUILD.gn
index 5f697a081c8..5633371102b 100644
--- a/chromium/third_party/webrtc/api/video/test/BUILD.gn
+++ b/chromium/third_party/webrtc/api/video/test/BUILD.gn
@@ -21,8 +21,8 @@ rtc_library("rtc_api_video_unittests") {
     "..:video_frame",
     "..:video_rtp_headers",
     "../../../test:test_support",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("mock_recordable_encoded_frame") {
diff --git a/chromium/third_party/webrtc/api/video/test/mock_recordable_encoded_frame.h b/chromium/third_party/webrtc/api/video/test/mock_recordable_encoded_frame.h
index 1788a493c6d..2178932d2a6 100644
--- a/chromium/third_party/webrtc/api/video/test/mock_recordable_encoded_frame.h
+++ b/chromium/third_party/webrtc/api/video/test/mock_recordable_encoded_frame.h
@@ -17,13 +17,18 @@
 namespace webrtc {
 class MockRecordableEncodedFrame : public RecordableEncodedFrame {
  public:
-  MOCK_CONST_METHOD0(encoded_buffer,
-                     rtc::scoped_refptr<const EncodedImageBufferInterface>());
-  MOCK_CONST_METHOD0(color_space, absl::optional<webrtc::ColorSpace>());
-  MOCK_CONST_METHOD0(codec, VideoCodecType());
-  MOCK_CONST_METHOD0(is_key_frame, bool());
-  MOCK_CONST_METHOD0(resolution, EncodedResolution());
-  MOCK_CONST_METHOD0(render_time, Timestamp());
+  MOCK_METHOD(rtc::scoped_refptr<const EncodedImageBufferInterface>,
+              encoded_buffer,
+              (),
+              (const, override));
+  MOCK_METHOD(absl::optional<webrtc::ColorSpace>,
+              color_space,
+              (),
+              (const, override));
+  MOCK_METHOD(VideoCodecType, codec, (), (const, override));
+  MOCK_METHOD(bool, is_key_frame, (), (const, override));
+  MOCK_METHOD(EncodedResolution, resolution, (), (const, override));
+  MOCK_METHOD(Timestamp, render_time, (), (const, override));
 };
 }  // namespace webrtc
 #endif  // API_VIDEO_TEST_MOCK_RECORDABLE_ENCODED_FRAME_H_
diff --git a/chromium/third_party/webrtc/api/video/video_adaptation_counters.cc b/chromium/third_party/webrtc/api/video/video_adaptation_counters.cc
index 25e0bee1ff8..df1769d5d46 100644
--- a/chromium/third_party/webrtc/api/video/video_adaptation_counters.cc
+++ b/chromium/third_party/webrtc/api/video/video_adaptation_counters.cc
@@ -10,6 +10,8 @@
 
 #include "api/video/video_adaptation_counters.h"
 
+#include "rtc_base/strings/string_builder.h"
+
 namespace webrtc {
 
 bool VideoAdaptationCounters::operator==(
@@ -30,4 +32,11 @@ VideoAdaptationCounters VideoAdaptationCounters::operator+(
       fps_adaptations + other.fps_adaptations);
 }
 
+std::string VideoAdaptationCounters::ToString() const {
+  rtc::StringBuilder ss;
+  ss << "{ res=" << resolution_adaptations << " fps=" << fps_adaptations
+     << " }";
+  return ss.Release();
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/video/video_adaptation_counters.h b/chromium/third_party/webrtc/api/video/video_adaptation_counters.h
index eff0baaa21f..2dea902f2f1 100644
--- a/chromium/third_party/webrtc/api/video/video_adaptation_counters.h
+++ b/chromium/third_party/webrtc/api/video/video_adaptation_counters.h
@@ -11,6 +11,8 @@
 #ifndef API_VIDEO_VIDEO_ADAPTATION_COUNTERS_H_
 #define API_VIDEO_VIDEO_ADAPTATION_COUNTERS_H_
 
+#include <string>
+
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -33,6 +35,8 @@ struct VideoAdaptationCounters {
 
   VideoAdaptationCounters operator+(const VideoAdaptationCounters& other) const;
 
+  std::string ToString() const;
+
   int resolution_adaptations;
   int fps_adaptations;
 };
diff --git a/chromium/third_party/webrtc/api/video/video_frame_marking.h b/chromium/third_party/webrtc/api/video/video_frame_marking.h
deleted file mode 100644
index 2a34852f1d6..00000000000
--- a/chromium/third_party/webrtc/api/video/video_frame_marking.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- *  Copyright (c) 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.
- */
-
-#ifndef API_VIDEO_VIDEO_FRAME_MARKING_H_
-#define API_VIDEO_VIDEO_FRAME_MARKING_H_
-
-namespace webrtc {
-
-struct FrameMarking {
-  bool start_of_frame;
-  bool end_of_frame;
-  bool independent_frame;
-  bool discardable_frame;
-  bool base_layer_sync;
-  uint8_t temporal_id;
-  uint8_t layer_id;
-  uint8_t tl0_pic_idx;
-};
-
-}  // namespace webrtc
-
-#endif  // API_VIDEO_VIDEO_FRAME_MARKING_H_
diff --git a/chromium/third_party/webrtc/api/video/video_frame_metadata.cc b/chromium/third_party/webrtc/api/video/video_frame_metadata.cc
new file mode 100644
index 00000000000..df82875eb95
--- /dev/null
+++ b/chromium/third_party/webrtc/api/video/video_frame_metadata.cc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#include "api/video/video_frame_metadata.h"
+
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+
+namespace webrtc {
+
+VideoFrameMetadata::VideoFrameMetadata(const RTPVideoHeader& header)
+    : width_(header.width), height_(header.height) {
+  if (header.generic) {
+    frame_id_ = header.generic->frame_id;
+    spatial_index_ = header.generic->spatial_index;
+    temporal_index_ = header.generic->temporal_index;
+    frame_dependencies_ = header.generic->dependencies;
+    decode_target_indications_ = header.generic->decode_target_indications;
+  }
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/video/video_frame_metadata.h b/chromium/third_party/webrtc/api/video/video_frame_metadata.h
new file mode 100644
index 00000000000..2e9309841b0
--- /dev/null
+++ b/chromium/third_party/webrtc/api/video/video_frame_metadata.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright 2020 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.
+ */
+
+#ifndef API_VIDEO_VIDEO_FRAME_METADATA_H_
+#define API_VIDEO_VIDEO_FRAME_METADATA_H_
+
+#include <cstdint>
+
+#include "absl/container/inlined_vector.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+
+namespace webrtc {
+
+struct RTPVideoHeader;
+
+// A subset of metadata from the RTP video header, exposed in insertable streams
+// API.
+class VideoFrameMetadata {
+ public:
+  explicit VideoFrameMetadata(const RTPVideoHeader& header);
+  VideoFrameMetadata(const VideoFrameMetadata&) = default;
+  VideoFrameMetadata& operator=(const VideoFrameMetadata&) = default;
+
+  uint16_t GetWidth() const { return width_; }
+  uint16_t GetHeight() const { return height_; }
+  absl::optional<int64_t> GetFrameId() const { return frame_id_; }
+  int GetSpatialIndex() const { return spatial_index_; }
+  int GetTemporalIndex() const { return temporal_index_; }
+
+  rtc::ArrayView<const int64_t> GetFrameDependencies() const {
+    return frame_dependencies_;
+  }
+
+  rtc::ArrayView<const DecodeTargetIndication> GetDecodeTargetIndications()
+      const {
+    return decode_target_indications_;
+  }
+
+ private:
+  int16_t width_;
+  int16_t height_;
+  absl::optional<int64_t> frame_id_;
+  int spatial_index_ = 0;
+  int temporal_index_ = 0;
+  absl::InlinedVector<int64_t, 5> frame_dependencies_;
+  absl::InlinedVector<DecodeTargetIndication, 10> decode_target_indications_;
+};
+}  // namespace webrtc
+
+#endif  // API_VIDEO_VIDEO_FRAME_METADATA_H_
diff --git a/chromium/third_party/webrtc/api/video/video_frame_metadata_unittest.cc b/chromium/third_party/webrtc/api/video/video_frame_metadata_unittest.cc
new file mode 100644
index 00000000000..7a808e1ea9b
--- /dev/null
+++ b/chromium/third_party/webrtc/api/video/video_frame_metadata_unittest.cc
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#include "api/video/video_frame_metadata.h"
+
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+
+TEST(VideoFrameMetadata, GetWidthReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  video_header.width = 1280u;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetWidth(), video_header.width);
+}
+
+TEST(VideoFrameMetadata, GetHeightReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  video_header.height = 720u;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetHeight(), video_header.height);
+}
+
+TEST(VideoFrameMetadata, GetFrameIdReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.frame_id = 10;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetFrameId().value(), 10);
+}
+
+TEST(VideoFrameMetadata, HasNoFrameIdForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_EQ(metadata.GetFrameId(), absl::nullopt);
+}
+
+TEST(VideoFrameMetadata, GetSpatialIndexReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.spatial_index = 2;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetSpatialIndex(), 2);
+}
+
+TEST(VideoFrameMetadata, SpatialIndexIsZeroForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_EQ(metadata.GetSpatialIndex(), 0);
+}
+
+TEST(VideoFrameMetadata, GetTemporalIndexReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.temporal_index = 3;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetTemporalIndex(), 3);
+}
+
+TEST(VideoFrameMetadata, TemporalIndexIsZeroForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_EQ(metadata.GetTemporalIndex(), 0);
+}
+
+TEST(VideoFrameMetadata, GetFrameDependenciesReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.dependencies = {5, 6, 7};
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_THAT(metadata.GetFrameDependencies(), ElementsAre(5, 6, 7));
+}
+
+TEST(VideoFrameMetadata, FrameDependencyVectorIsEmptyForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_THAT(metadata.GetFrameDependencies(), IsEmpty());
+}
+
+TEST(VideoFrameMetadata, GetDecodeTargetIndicationsReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch};
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_THAT(metadata.GetDecodeTargetIndications(),
+              ElementsAre(DecodeTargetIndication::kSwitch));
+}
+
+TEST(VideoFrameMetadata,
+     DecodeTargetIndicationsVectorIsEmptyForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_THAT(metadata.GetDecodeTargetIndications(), IsEmpty());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/video/video_stream_encoder_interface.h b/chromium/third_party/webrtc/api/video/video_stream_encoder_interface.h
index 8e1df0f858a..d8dd8e15994 100644
--- a/chromium/third_party/webrtc/api/video/video_stream_encoder_interface.h
+++ b/chromium/third_party/webrtc/api/video/video_stream_encoder_interface.h
@@ -13,8 +13,10 @@
 
 #include <vector>
 
+#include "api/adaptation/resource.h"
 #include "api/fec_controller_override.h"
 #include "api/rtp_parameters.h"  // For DegradationPreference.
+#include "api/scoped_refptr.h"
 #include "api/units/data_rate.h"
 #include "api/video/video_bitrate_allocator.h"
 #include "api/video/video_sink_interface.h"
@@ -49,6 +51,15 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
         int min_transmit_bitrate_bps) = 0;
   };
 
+  // If the resource is overusing, the VideoStreamEncoder will try to reduce
+  // resolution or frame rate until no resource is overusing.
+  // TODO(https://crbug.com/webrtc/11565): When the ResourceAdaptationProcessor
+  // is moved to Call this method could be deleted altogether in favor of
+  // Call-level APIs only.
+  virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
+  virtual std::vector<rtc::scoped_refptr<Resource>>
+  GetAdaptationResources() = 0;
+
   // Sets the source that will provide video frames to the VideoStreamEncoder's
   // OnFrame method. |degradation_preference| control whether or not resolution
   // or frame rate may be reduced. The VideoStreamEncoder registers itself with