Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED

Bug: None
Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30283}

api/video/BUILD.gn

@@ -58,9 +58,7 @@ rtc_library("video_frame") {
 
 rtc_source_set("recordable_encoded_frame") {
   visibility = [ "*" ]
-  sources = [
+  sources = [ "recordable_encoded_frame.h" ]
-    "recordable_encoded_frame.h",
-  ]
 
   deps = [
     ":encoded_image",
@@ -75,9 +73,7 @@ rtc_source_set("recordable_encoded_frame") {
 
 rtc_source_set("video_frame_type") {
   visibility = [ "*" ]
-  sources = [
+  sources = [ "video_frame_type.h" ]
-    "video_frame_type.h",
-  ]
 }
 
 rtc_library("video_frame_i420") {
@@ -146,16 +142,12 @@ rtc_library("encoded_frame") {
     "encoded_frame.h",
   ]
 
-  deps = [
+  deps = [ "../../modules/video_coding:encoded_frame" ]
-    "../../modules/video_coding:encoded_frame",
-  ]
 }
 
 rtc_source_set("video_codec_constants") {
   visibility = [ "*" ]
-  sources = [
+  sources = [ "video_codec_constants.h" ]
-    "video_codec_constants.h",
-  ]
   deps = []
 }
 
@@ -189,9 +181,7 @@ rtc_library("video_bitrate_allocator") {
 
 rtc_source_set("video_bitrate_allocator_factory") {
   visibility = [ "*" ]
-  sources = [
+  sources = [ "video_bitrate_allocator_factory.h" ]
-    "video_bitrate_allocator_factory.h",
-  ]
   deps = [
     ":video_bitrate_allocator",
     "../../rtc_base:rtc_base_approved",
@@ -201,9 +191,7 @@ rtc_source_set("video_bitrate_allocator_factory") {
 
 rtc_source_set("video_stream_decoder") {
   visibility = [ "*" ]
-  sources = [
+  sources = [ "video_stream_decoder.h" ]
-    "video_stream_decoder.h",
-  ]
 
   deps = [
     ":encoded_frame",
@@ -288,17 +276,15 @@ rtc_library("builtin_video_bitrate_allocator_factory") {
     "../../media:rtc_media_base",
     "../../modules/video_coding:video_coding_utility",
     "../../modules/video_coding:webrtc_vp9_helpers",
-    "../../rtc_base/system:fallthrough",
     "../video_codecs:video_codecs_api",
+    "//third_party/abseil-cpp/absl/base:core_headers",
   ]
 }
 
 if (rtc_include_tests) {
   rtc_library("video_unittests") {
     testonly = true
-    sources = [
+    sources = [ "video_stream_decoder_create_unittest.cc" ]
-      "video_stream_decoder_create_unittest.cc",
-    ]
     deps = [
       ":video_stream_decoder_create",
       "../../test:test_support",

api/video/builtin_video_bitrate_allocator_factory.cc

@@ -12,12 +12,12 @@
 
 #include <memory>
 
+#include "absl/base/macros.h"
 #include "api/video/video_bitrate_allocator.h"
 #include "api/video_codecs/video_codec.h"
 #include "modules/video_coding/codecs/vp9/svc_rate_allocator.h"
 #include "modules/video_coding/utility/default_video_bitrate_allocator.h"
 #include "modules/video_coding/utility/simulcast_rate_allocator.h"
-#include "rtc_base/system/fallthrough.h"
 
 namespace webrtc {
 
@@ -34,7 +34,7 @@ class BuiltinVideoBitrateAllocatorFactory
     std::unique_ptr<VideoBitrateAllocator> rate_allocator;
     switch (codec.codecType) {
       case kVideoCodecVP8:
-        RTC_FALLTHROUGH();
+        ABSL_FALLTHROUGH_INTENDED;
       case kVideoCodecH264:
         rate_allocator.reset(new SimulcastRateAllocator(codec));
         break;

api/video_codecs/BUILD.gn

@@ -58,12 +58,8 @@ rtc_library("video_codecs_api") {
 
 rtc_source_set("bitstream_parser_api") {
   visibility = [ "*" ]
-  sources = [
+  sources = [ "bitstream_parser.h" ]
-    "bitstream_parser.h",
+  deps = [ "..:array_view" ]
-  ]
-  deps = [
-    "..:array_view",
-  ]
 }
 
 rtc_library("builtin_video_decoder_factory") {
@@ -143,13 +139,13 @@ rtc_library("rtc_software_fallback_wrappers") {
     "../../modules/video_coding:video_codec_interface",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
-    "../../rtc_base/system:fallthrough",
     "../../rtc_base/system:rtc_export",
     "../../system_wrappers:field_trial",
     "../video:encoded_image",
     "../video:video_bitrate_allocation",
     "../video:video_frame",
     "../video:video_rtp_headers",
+    "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }

api/video_codecs/video_decoder_software_fallback_wrapper.cc

@@ -16,12 +16,12 @@
 #include <string>
 #include <utility>
 
+#include "absl/base/macros.h"
 #include "api/video/encoded_image.h"
 #include "api/video_codecs/video_codec.h"
 #include "modules/video_coding/include/video_error_codes.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/system/fallthrough.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"
 
@@ -166,7 +166,7 @@ int32_t VideoDecoderSoftwareFallbackWrapper::Decode(
       }
 
       // Fallback decoder initialized, fall-through.
-      RTC_FALLTHROUGH();
+      ABSL_FALLTHROUGH_INTENDED;
     }
     case DecoderType::kFallback:
       return fallback_decoder_->Decode(input_image, missing_frames,

modules/audio_coding/BUILD.gn

@@ -16,12 +16,8 @@ visibility = [ ":*" ]
 
 rtc_source_set("audio_coding_module_typedefs") {
   visibility += [ "*" ]
-  sources = [
+  sources = [ "include/audio_coding_module_typedefs.h" ]
-    "include/audio_coding_module_typedefs.h",
+  deps = [ "../../rtc_base:deprecation" ]
-  ]
-  deps = [
-    "../../rtc_base:deprecation",
-  ]
 }
 
 rtc_library("audio_coding") {
@@ -144,9 +140,7 @@ rtc_library("g711") {
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
   ]
-  public_deps = [
+  public_deps = [ ":g711_c" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":g711_c",
-  ]
 }
 
 rtc_library("g711_c") {
@@ -155,9 +149,7 @@ rtc_library("g711_c") {
     "codecs/g711/g711_interface.c",
     "codecs/g711/g711_interface.h",
   ]
-  deps = [
+  deps = [ "../third_party/g711:g711_3p" ]
-    "../third_party/g711:g711_3p",
-  ]
 }
 
 rtc_library("g722") {
@@ -178,9 +170,7 @@ rtc_library("g722") {
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
   ]
-  public_deps = [
+  public_deps = [ ":g722_c" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":g722_c",
-  ]
 }
 
 rtc_library("g722_c") {
@@ -189,9 +179,7 @@ rtc_library("g722_c") {
     "codecs/g722/g722_interface.c",
     "codecs/g722/g722_interface.h",
   ]
-  deps = [
+  deps = [ "../third_party/g722:g722_3p" ]
-    "../third_party/g722:g722_3p",
-  ]
 }
 
 rtc_library("ilbc") {
@@ -213,9 +201,7 @@ rtc_library("ilbc") {
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
   ]
-  public_deps = [
+  public_deps = [ ":ilbc_c" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":ilbc_c",
-  ]
 }
 
 rtc_library("ilbc_c") {
@@ -406,15 +392,11 @@ rtc_library("isac") {
     ":isac_common",
     "../../api/audio_codecs:audio_codecs_api",
   ]
-  public_deps = [
+  public_deps = [ ":isac_c" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":isac_c",
-  ]
 }
 
 rtc_source_set("isac_bwinfo") {
-  sources = [
+  sources = [ "codecs/isac/bandwidth_info.h" ]
-    "codecs/isac/bandwidth_info.h",
-  ]
   deps = []
 }
 
@@ -517,9 +499,7 @@ rtc_library("isac_fix") {
     "../../common_audio",
     "../../system_wrappers",
   ]
-  public_deps = [
+  public_deps = [ ":isac_fix_c" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":isac_fix_c",
-  ]
 
   if (rtc_build_with_neon) {
     deps += [ ":isac_neon" ]
@@ -618,9 +598,7 @@ rtc_library("isac_fix_c") {
     "../third_party/fft",
   ]
 
-  public_deps = [
+  public_deps = [ ":isac_fix_common" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":isac_fix_common",
-  ]
 
   if (rtc_build_with_neon) {
     deps += [ ":isac_neon" ]
@@ -713,9 +691,7 @@ rtc_library("pcm16b") {
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
   ]
-  public_deps = [
+  public_deps = [ ":pcm16b_c" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":pcm16b_c",
-  ]
 }
 
 rtc_library("pcm16b_c") {
@@ -768,9 +744,7 @@ rtc_library("webrtc_opus") {
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
-  public_deps = [  # no-presubmit-check TODO(webrtc:8603)
+  public_deps = [ ":webrtc_opus_wrapper" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":webrtc_opus_wrapper",
-  ]
 
   defines = audio_codec_defines
 
@@ -806,9 +780,7 @@ rtc_library("webrtc_multiopus") {
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
-  public_deps = [  # no-presubmit-check TODO(webrtc:8603)
+  public_deps = [ ":webrtc_opus_wrapper" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":webrtc_opus_wrapper",
-  ]
 
   defines = audio_codec_defines
 
@@ -830,9 +802,7 @@ rtc_library("webrtc_opus_wrapper") {
   defines = audio_coding_defines
 
   if (rtc_build_opus) {
-    public_deps = [
+    public_deps = [ rtc_opus_dir ]  # no-presubmit-check TODO(webrtc:8603)
-      rtc_opus_dir,
-    ]
   } else if (build_with_mozilla) {
     include_dirs = [ getenv("DIST") + "/include/opus" ]
   }
@@ -848,17 +818,13 @@ rtc_library("webrtc_opus_wrapper") {
 if (rtc_enable_protobuf) {
   proto_library("ana_debug_dump_proto") {
     visibility += webrtc_default_visibility
-    sources = [
+    sources = [ "audio_network_adaptor/debug_dump.proto" ]
-      "audio_network_adaptor/debug_dump.proto",
-    ]
     link_deps = [ ":ana_config_proto" ]
     proto_out_dir = "modules/audio_coding/audio_network_adaptor"
   }
   proto_library("ana_config_proto") {
     visibility += [ "*" ]
-    sources = [
+    sources = [ "audio_network_adaptor/config.proto" ]
-      "audio_network_adaptor/config.proto",
-    ]
     proto_out_dir = "modules/audio_coding/audio_network_adaptor"
   }
 }
@@ -869,9 +835,7 @@ rtc_library("audio_network_adaptor_config") {
     "audio_network_adaptor/audio_network_adaptor_config.cc",
     "audio_network_adaptor/include/audio_network_adaptor_config.h",
   ]
-  deps = [
+  deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
-    "//third_party/abseil-cpp/absl/types:optional",
-  ]
 }
 
 rtc_library("audio_network_adaptor") {
@@ -901,9 +865,7 @@ rtc_library("audio_network_adaptor") {
     "audio_network_adaptor/util/threshold_curve.h",
   ]
 
-  public_deps = [
+  public_deps = [ ":audio_network_adaptor_config" ]  # no-presubmit-check TODO(webrtc:8603)
-    ":audio_network_adaptor_config",
-  ]
 
   deps = [
     "../../api/audio_codecs:audio_codecs_api",
@@ -1016,7 +978,6 @@ rtc_library("neteq") {
     "../../rtc_base:safe_minmax",
     "../../rtc_base:sanitizer",
     "../../rtc_base/experiments:field_trial_parser",
-    "../../rtc_base/system:fallthrough",
     "../../system_wrappers",
     "../../system_wrappers:field_trial",
     "../../system_wrappers:metrics",
@@ -1199,9 +1160,7 @@ if (rtc_enable_protobuf) {
       "../rtp_rtcp:rtp_rtcp_format",
       "//third_party/abseil-cpp/absl/types:optional",
     ]
-    public_deps = [
+    public_deps = [ "../../logging:rtc_event_log_proto" ]  # no-presubmit-check TODO(webrtc:8603)
-      "../../logging:rtc_event_log_proto",
-    ]
   }
 
   # Only used for test purpose. Since we want to use it from chromium
@@ -1209,9 +1168,7 @@ if (rtc_enable_protobuf) {
   # under rtc_include_tests.
   proto_library("neteq_unittest_proto") {
     testonly = true
-    sources = [
+    sources = [ "neteq/neteq_unittest.proto" ]
-      "neteq/neteq_unittest.proto",
-    ]
     proto_out_dir = "modules/audio_coding/neteq"
   }
 }
@@ -1480,17 +1437,13 @@ if (rtc_include_tests) {
     bundle_data("audio_decoder_unittests_bundle_data") {
       testonly = true
       sources = audio_decoder_unittests_resources
-      outputs = [
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
-        "{{bundle_resources_dir}}/{{source_file_part}}",
-      ]
     }
   }
 
   rtc_test("audio_decoder_unittests") {
     testonly = true
-    sources = [
+    sources = [ "neteq/audio_decoder_unittest.cc" ]
-      "neteq/audio_decoder_unittest.cc",
-    ]
 
     defines = neteq_defines
 
@@ -1564,9 +1517,7 @@ if (rtc_include_tests) {
         "//third_party/abseil-cpp/absl/strings",
         "//third_party/abseil-cpp/absl/types:optional",
       ]
-      sources = [
+      sources = [ "neteq/tools/neteq_rtpplay.cc" ]
-        "neteq/tools/neteq_rtpplay.cc",
-      ]
     }
   }
 
@@ -1580,18 +1531,14 @@ if (rtc_include_tests) {
     bundle_data("audio_codec_speed_tests_data") {
       testonly = true
       sources = audio_codec_speed_tests_resources
-      outputs = [
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
-        "{{bundle_resources_dir}}/{{source_file_part}}",
-      ]
     }
   }
 
   rtc_test("audio_codec_speed_tests") {
     testonly = true
     defines = []
-    deps = [
+    deps = [ "../../test:fileutils" ]
-      "../../test:fileutils",
-    ]
     sources = [
       "codecs/isac/fix/test/isac_speed_test.cc",
       "codecs/opus/opus_speed_test.cc",
@@ -1688,9 +1635,7 @@ if (rtc_include_tests) {
              "../../rtc_base:safe_conversions",
            ]
 
-    sources = [
+    sources = [ "neteq/tools/rtp_encode.cc" ]
-      "neteq/tools/rtp_encode.cc",
-    ]
 
     defines = audio_coding_defines
   }
@@ -1704,9 +1649,7 @@ if (rtc_include_tests) {
              "../../rtc_base:rtc_base_approved",
            ]
 
-    sources = [
+    sources = [ "neteq/tools/rtp_jitter.cc" ]
-      "neteq/tools/rtp_jitter.cc",
-    ]
 
     defines = audio_coding_defines
   }
@@ -1714,9 +1657,7 @@ if (rtc_include_tests) {
   rtc_executable("rtpcat") {
     testonly = true
 
-    sources = [
+    sources = [ "neteq/tools/rtpcat.cc" ]
-      "neteq/tools/rtpcat.cc",
-    ]
 
     deps = [
       "../../rtc_base:checks",
@@ -1729,9 +1670,7 @@ if (rtc_include_tests) {
   rtc_executable("rtp_analyze") {
     testonly = true
 
-    sources = [
+    sources = [ "neteq/tools/rtp_analyze.cc" ]
-      "neteq/tools/rtp_analyze.cc",
-    ]
 
     deps = [
       ":neteq",
@@ -1746,9 +1685,7 @@ if (rtc_include_tests) {
   rtc_executable("neteq_opus_quality_test") {
     testonly = true
 
-    sources = [
+    sources = [ "neteq/test/neteq_opus_quality_test.cc" ]
-      "neteq/test/neteq_opus_quality_test.cc",
-    ]
 
     deps = [
       ":neteq",
@@ -1765,9 +1702,7 @@ if (rtc_include_tests) {
   rtc_executable("neteq_speed_test") {
     testonly = true
 
-    sources = [
+    sources = [ "neteq/test/neteq_speed_test.cc" ]
-      "neteq/test/neteq_speed_test.cc",
-    ]
 
     deps = [
       ":neteq",
@@ -1782,9 +1717,7 @@ if (rtc_include_tests) {
   rtc_executable("neteq_ilbc_quality_test") {
     testonly = true
 
-    sources = [
+    sources = [ "neteq/test/neteq_ilbc_quality_test.cc" ]
-      "neteq/test/neteq_ilbc_quality_test.cc",
-    ]
 
     deps = [
       ":ilbc",
@@ -1803,9 +1736,7 @@ if (rtc_include_tests) {
   rtc_executable("neteq_isac_quality_test") {
     testonly = true
 
-    sources = [
+    sources = [ "neteq/test/neteq_isac_quality_test.cc" ]
-      "neteq/test/neteq_isac_quality_test.cc",
-    ]
 
     deps = [
       ":isac_fix",
@@ -1821,9 +1752,7 @@ if (rtc_include_tests) {
   rtc_executable("neteq_pcmu_quality_test") {
     testonly = true
 
-    sources = [
+    sources = [ "neteq/test/neteq_pcmu_quality_test.cc" ]
-      "neteq/test/neteq_pcmu_quality_test.cc",
-    ]
 
     deps = [
       ":g711",
@@ -1841,9 +1770,7 @@ if (rtc_include_tests) {
   rtc_executable("neteq_pcm16b_quality_test") {
     testonly = true
 
-    sources = [
+    sources = [ "neteq/test/neteq_pcm16b_quality_test.cc" ]
-      "neteq/test/neteq_pcm16b_quality_test.cc",
-    ]
 
     deps = [
       ":neteq",
@@ -1861,9 +1788,7 @@ if (rtc_include_tests) {
   rtc_executable("isac_fix_test") {
     testonly = true
 
-    sources = [
+    sources = [ "codecs/isac/fix/test/kenny.cc" ]
-      "codecs/isac/fix/test/kenny.cc",
-    ]
 
     deps = [
       ":isac_fix",
@@ -1871,9 +1796,7 @@ if (rtc_include_tests) {
       "../../test:test_support",
     ]
 
-    data = [
+    data = [ "../../resources/speech_and_misc_wb.pcm" ]
-      "../../resources/speech_and_misc_wb.pcm",
-    ]
   }
 
   rtc_library("isac_test_util") {
@@ -1887,9 +1810,7 @@ if (rtc_include_tests) {
   rtc_executable("isac_test") {
     testonly = true
 
-    sources = [
+    sources = [ "codecs/isac/main/test/simpleKenny.c" ]
-      "codecs/isac/main/test/simpleKenny.c",
-    ]
 
     deps = [
       ":isac",
@@ -1901,33 +1822,23 @@ if (rtc_include_tests) {
   rtc_executable("g711_test") {
     testonly = true
 
-    sources = [
+    sources = [ "codecs/g711/test/testG711.cc" ]
-      "codecs/g711/test/testG711.cc",
-    ]
 
-    deps = [
+    deps = [ ":g711" ]
-      ":g711",
-    ]
   }
 
   rtc_executable("g722_test") {
     testonly = true
 
-    sources = [
+    sources = [ "codecs/g722/test/testG722.cc" ]
-      "codecs/g722/test/testG722.cc",
-    ]
 
-    deps = [
+    deps = [ ":g722" ]
-      ":g722",
-    ]
   }
 
   rtc_executable("isac_api_test") {
     testonly = true
 
-    sources = [
+    sources = [ "codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc" ]
-      "codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc",
-    ]
 
     deps = [
       ":isac",
@@ -1939,9 +1850,7 @@ if (rtc_include_tests) {
   rtc_executable("isac_switch_samprate_test") {
     testonly = true
 
-    sources = [
+    sources = [ "codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc" ]
-      "codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc",
-    ]
 
     deps = [
       ":isac",
@@ -1954,21 +1863,15 @@ if (rtc_include_tests) {
   rtc_executable("ilbc_test") {
     testonly = true
 
-    sources = [
+    sources = [ "codecs/ilbc/test/iLBC_test.c" ]
-      "codecs/ilbc/test/iLBC_test.c",
-    ]
 
-    deps = [
+    deps = [ ":ilbc" ]
-      ":ilbc",
-    ]
   }
 
   rtc_executable("webrtc_opus_fec_test") {
     testonly = true
 
-    sources = [
+    sources = [ "codecs/opus/opus_fec_test.cc" ]
-      "codecs/opus/opus_fec_test.cc",
-    ]
 
     deps = [
       ":webrtc_opus",
@@ -2147,12 +2050,8 @@ if (rtc_include_tests) {
 # TODO(kwiberg): Remove this.
 rtc_source_set("audio_decoder_interface") {
   visibility += [ "*" ]
-  sources = [
+  sources = [ "codecs/audio_decoder.h" ]
-    "codecs/audio_decoder.h",
+  deps = [ "../../api/audio_codecs:audio_codecs_api" ]
-  ]
-  deps = [
-    "../../api/audio_codecs:audio_codecs_api",
-  ]
 }
 
 # For backwards compatibility only! Use
@@ -2160,10 +2059,6 @@ rtc_source_set("audio_decoder_interface") {
 # TODO(ossu): Remove this.
 rtc_source_set("audio_encoder_interface") {
   visibility += [ "*" ]
-  sources = [
+  sources = [ "codecs/audio_encoder.h" ]
-    "codecs/audio_encoder.h",
+  deps = [ "../../api/audio_codecs:audio_codecs_api" ]
-  ]
-  deps = [
-    "../../api/audio_codecs:audio_codecs_api",
-  ]
 }

modules/congestion_controller/bbr/BUILD.gn

@@ -36,8 +36,8 @@ rtc_library("bbr_controller") {
     "../../../rtc_base:checks",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/experiments:field_trial_parser",
-    "../../../rtc_base/system:fallthrough",
     "../../../system_wrappers:field_trial",
+    "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -77,12 +77,8 @@ rtc_library("data_transfer_tracker") {
 
 rtc_source_set("packet_number_indexed_queue") {
   visibility = [ ":*" ]
-  sources = [
+  sources = [ "packet_number_indexed_queue.h" ]
-    "packet_number_indexed_queue.h",
+  deps = [ "../../../rtc_base:checks" ]
-  ]
-  deps = [
-    "../../../rtc_base:checks",
-  ]
 }
 
 rtc_library("loss_rate_filter") {
@@ -91,9 +87,7 @@ rtc_library("loss_rate_filter") {
     "loss_rate_filter.cc",
     "loss_rate_filter.h",
   ]
-  deps = [
+  deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
-    "//third_party/abseil-cpp/absl/types:optional",
-  ]
 }
 rtc_library("rtt_stats") {
   visibility = [ ":*" ]
@@ -110,9 +104,7 @@ rtc_library("rtt_stats") {
 }
 rtc_source_set("windowed_filter") {
   visibility = [ ":*" ]
-  sources = [
+  sources = [ "windowed_filter.h" ]
-    "windowed_filter.h",
-  ]
 }
 if (rtc_include_tests) {
   rtc_library("bbr_unittests") {

modules/congestion_controller/bbr/bbr_network_controller.cc

@@ -15,9 +15,9 @@
 #include <string>
 #include <vector>
 
+#include "absl/base/macros.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/system/fallthrough.h"
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
@@ -784,7 +784,7 @@ void BbrNetworkController::UpdateRecoveryState(int64_t last_acked_packet,
       if (is_round_start) {
         recovery_state_ = GROWTH;
       }
-      RTC_FALLTHROUGH();
+      ABSL_FALLTHROUGH_INTENDED;
     case GROWTH:
       // Exit recovery if appropriate.
       if (!has_losses &&

modules/rtp_rtcp/BUILD.gn

@@ -277,13 +277,13 @@ rtc_library("rtp_rtcp") {
     "../../rtc_base:rtc_numerics",
     "../../rtc_base:safe_minmax",
     "../../rtc_base/synchronization:sequence_checker",
-    "../../rtc_base/system:fallthrough",
     "../../rtc_base/time:timestamp_extrapolator",
     "../../system_wrappers",
     "../../system_wrappers:metrics",
     "../remote_bitrate_estimator",
     "../video_coding:codec_globals_headers",
     "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/container:inlined_vector",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
@@ -404,9 +404,7 @@ if (rtc_include_tests) {
   rtc_library("rtp_rtcp_modules_tests") {
     testonly = true
 
-    sources = [
+    sources = [ "test/testFec/test_fec.cc" ]
-      "test/testFec/test_fec.cc",
-    ]
     deps = [
       ":rtp_rtcp",
       ":rtp_rtcp_format",

modules/rtp_rtcp/source/rtp_format_h264.cc

@@ -30,7 +30,6 @@
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/system/fallthrough.h"
 
 namespace webrtc {
 namespace {

modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc

@@ -15,6 +15,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/base/macros.h"
 #include "absl/types/optional.h"
 #include "absl/types/variant.h"
 #include "common_video/h264/h264_common.h"
@@ -26,7 +27,6 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/system/fallthrough.h"
 
 namespace webrtc {
 namespace {
@@ -197,7 +197,7 @@ absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
       case H264::NaluType::kIdr:
         parsed_payload->video_header.frame_type =
             VideoFrameType::kVideoFrameKey;
-        RTC_FALLTHROUGH();
+        ABSL_FALLTHROUGH_INTENDED;
       case H264::NaluType::kSlice: {
         absl::optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
             &payload_data[start_offset], end_offset - start_offset);

modules/video_coding/BUILD.gn

@@ -152,7 +152,6 @@ rtc_library("video_coding") {
     "../../rtc_base/experiments:rate_control_settings",
     "../../rtc_base/experiments:rtt_mult_experiment",
     "../../rtc_base/synchronization:sequence_checker",
-    "../../rtc_base/system:fallthrough",
     "../../rtc_base/task_utils:repeating_task",
     "../../rtc_base/third_party/base64",
     "../../rtc_base/time:timestamp_extrapolator",
@@ -249,9 +248,7 @@ rtc_source_set("codec_globals_headers") {
     "codecs/vp9/include/vp9_globals.h",
   ]
 
-  deps = [
+  deps = [ "../../rtc_base:checks" ]
-    "../../rtc_base:checks",
-  ]
 }
 
 rtc_library("video_coding_utility") {
@@ -672,9 +669,7 @@ if (rtc_include_tests) {
       bundle_data("video_coding_modules_tests_resources_bundle_data") {
         testonly = true
         sources = video_coding_modules_tests_resources
-        outputs = [
+        outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
-          "{{bundle_resources_dir}}/{{source_file_part}}",
-        ]
       }
     }
   }

modules/video_coding/rtp_frame_reference_finder.cc

@@ -13,12 +13,12 @@
 #include <algorithm>
 #include <limits>
 
+#include "absl/base/macros.h"
 #include "absl/types/variant.h"
 #include "modules/video_coding/frame_object.h"
 #include "modules/video_coding/packet_buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/system/fallthrough.h"
 
 namespace webrtc {
 namespace video_coding {
@@ -78,7 +78,7 @@ void RtpFrameReferenceFinder::RetryStashedFrames() {
         case kHandOff:
           complete_frame = true;
           HandOffFrame(std::move(*frame_it));
-          RTC_FALLTHROUGH();
+          ABSL_FALLTHROUGH_INTENDED;
         case kDrop:
           frame_it = stashed_frames_.erase(frame_it);
       }

pc/BUILD.gn

@@ -13,9 +13,7 @@ if (is_android) {
 }
 
 group("pc") {
-  deps = [
+  deps = [ ":rtc_pc" ]
-    ":rtc_pc",
-  ]
 }
 
 config("rtc_pc_config") {
@@ -267,7 +265,6 @@ rtc_library("peerconnection") {
     "../rtc_base:safe_minmax",
     "../rtc_base:weak_ptr",
     "../rtc_base/experiments:field_trial_parser",
-    "../rtc_base/system:fallthrough",
     "../rtc_base/system:file_wrapper",
     "../rtc_base/system:rtc_export",
     "../rtc_base/third_party/base64",
@@ -375,9 +372,7 @@ if (rtc_include_tests) {
 
   rtc_library("peerconnection_perf_tests") {
     testonly = true
-    sources = [
+    sources = [ "peer_connection_rampup_tests.cc" ]
-      "peer_connection_rampup_tests.cc",
-    ]
     deps = [
       ":pc_test_utils",
       ":peerconnection_wrapper",

pc/peer_connection.cc

@@ -56,7 +56,6 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/string_encode.h"
 #include "rtc_base/strings/string_builder.h"
-#include "rtc_base/system/fallthrough.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/field_trial.h"

rtc_base/system/BUILD.gn

@@ -13,21 +13,11 @@ if (is_android) {
 }
 
 rtc_source_set("arch") {
-  sources = [
+  sources = [ "arch.h" ]
-    "arch.h",
-  ]
 }
 
 rtc_source_set("asm_defines") {
-  sources = [
+  sources = [ "asm_defines.h" ]
-    "asm_defines.h",
-  ]
-}
-
-rtc_source_set("fallthrough") {
-  sources = [
-    "fallthrough.h",
-  ]
 }
 
 rtc_library("file_wrapper") {
@@ -43,21 +33,15 @@ rtc_library("file_wrapper") {
 }
 
 rtc_source_set("ignore_warnings") {
-  sources = [
+  sources = [ "ignore_warnings.h" ]
-    "ignore_warnings.h",
-  ]
 }
 
 rtc_source_set("inline") {
-  sources = [
+  sources = [ "inline.h" ]
-    "inline.h",
-  ]
 }
 
 rtc_source_set("unused") {
-  sources = [
+  sources = [ "unused.h" ]
-    "unused.h",
-  ]
 }
 
 rtc_source_set("rtc_export") {
@@ -73,20 +57,14 @@ if (is_mac || is_ios) {
       "cocoa_threading.h",
       "cocoa_threading.mm",
     ]
-    deps = [
+    deps = [ "..:checks" ]
-      "..:checks",
-    ]
     libs = [ "Foundation.framework" ]
   }
 }
 
 rtc_source_set("thread_registry") {
-  sources = [
+  sources = [ "thread_registry.h" ]
-    "thread_registry.h",
+  deps = [ "..:rtc_base_approved" ]
-  ]
-  deps = [
-    "..:rtc_base_approved",
-  ]
   if (is_android && !build_with_chromium) {
     sources += [ "thread_registry.cc" ]
     deps += [
@@ -97,9 +75,7 @@ rtc_source_set("thread_registry") {
 }
 
 rtc_source_set("warn_current_thread_is_deadlocked") {
-  sources = [
+  sources = [ "warn_current_thread_is_deadlocked.h" ]
-    "warn_current_thread_is_deadlocked.h",
-  ]
   deps = []
   if (is_android && !build_with_chromium) {
     sources += [ "warn_current_thread_is_deadlocked.cc" ]

rtc_base/system/fallthrough.h

@@ -1,31 +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 RTC_BASE_SYSTEM_FALLTHROUGH_H_
-#define RTC_BASE_SYSTEM_FALLTHROUGH_H_
-
-// Macro to be used for switch-case fallthrough (required for enabling
-// -Wimplicit-fallthrough warning on Clang).
-
-// This macro definition must not be included from public headers! Because
-// clang's diagnostic checks if there's a macro expanding to
-// [[clang::fallthrough]] defined, and if so it suggests the first macro
-// expanding to it. So if this macro is included in a public header, clang may
-// suggest it instead of the client's own macro, which can cause confusion.
-
-#ifdef __clang__
-#define RTC_FALLTHROUGH() [[clang::fallthrough]]
-#else
-#define RTC_FALLTHROUGH() \
-  do {                    \
-  } while (0)
-#endif
-
-#endif  // RTC_BASE_SYSTEM_FALLTHROUGH_H_

sdk/BUILD.gn

@@ -41,9 +41,7 @@ rtc_library("media_constraints") {
 
 rtc_library("sdk_tests") {
   testonly = true
-  sources = [
+  sources = [ "media_constraints_unittest.cc" ]
-    "media_constraints_unittest.cc",
-  ]
   deps = [
     ":media_constraints",
     "../test:test_support",
@@ -84,9 +82,7 @@ if (is_ios || is_mac) {
   rtc_library("common_objc") {
     visibility = [ "*" ]
 
-    sources = [
+    sources = [ "objc/helpers/noop.mm" ]
-      "objc/helpers/noop.mm",
-    ]
 
     public_configs = [ ":common_config_objc" ]
 
@@ -243,13 +239,9 @@ if (is_ios || is_mac) {
       rtc_source_set("audio_session_observer") {
         visibility = [ ":*" ]
 
-        sources = [
+        sources = [ "objc/native/src/audio/audio_session_observer.h" ]
-          "objc/native/src/audio/audio_session_observer.h",
-        ]
 
-        deps = [
+        deps = [ "../rtc_base" ]
-          "../rtc_base",
-        ]
       }
 
       rtc_library("audio_device") {
@@ -278,9 +270,9 @@ if (is_ios || is_mac) {
           "../modules/audio_device:audio_device_generic",
           "../rtc_base",
           "../rtc_base:checks",
-          "../rtc_base/system:fallthrough",
           "../system_wrappers:field_trial",
           "../system_wrappers:metrics",
+          "//third_party/abseil-cpp/absl/base:core_headers",
         ]
 
         libs = [ "AudioToolbox.framework" ]
@@ -542,9 +534,7 @@ if (is_ios || is_mac) {
     # TODO(bugs.webrtc.org/9627): Remove this target.
     rtc_library("videocapturebase_objc") {
       visibility = [ "*" ]
-      sources = [
+      sources = [ "objc/helpers/noop.mm" ]
-        "objc/helpers/noop.mm",
-      ]
 
       configs += [ "..:common_objc" ]
 
@@ -700,9 +690,7 @@ if (is_ios || is_mac) {
       ]
       defines = [ "HAVE_NO_MEDIA" ]
 
-      sources = [
+      sources = [ "objc/helpers/noop.mm" ]
-        "objc/helpers/noop.mm",
-      ]
 
       public_configs = [ ":common_config_objc" ]
 
@@ -734,16 +722,12 @@ if (is_ios || is_mac) {
     # TODO(bugs.webrtc.org/9627): Remove, targets should depend on base_objc.
     rtc_library("videorenderer_objc") {
       visibility = [ "*" ]
-      sources = [
+      sources = [ "objc/helpers/noop.mm" ]
-        "objc/helpers/noop.mm",
-      ]
 
       configs += [ "..:common_objc" ]
       public_configs = [ ":common_config_objc" ]
 
-      deps = [
+      deps = [ ":base_objc" ]
-        ":base_objc",
-      ]
     }
 
     rtc_library("videorendereradapter_objc") {
@@ -1152,17 +1136,13 @@ if (is_ios || is_mac) {
             # Sample video taken from https://media.xiph.org/video/derf/
             "objc/unittests/foreman.mp4",
           ]
-          outputs = [
+          outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
-            "{{bundle_resources_dir}}/{{source_file_part}}",
-          ]
         }
 
         # These tests use static linking.
         rtc_ios_xctest_test("sdk_unittests") {
           info_plist = "//test/ios/Info.plist"
-          sources = [
+          sources = [ "objc/unittests/main.mm" ]
-            "objc/unittests/main.mm",
-          ]
 
           extra_substitutions = [ "GTEST_BUNDLE_ID_SUFFIX=generic-unit-test" ]
           deps = [
@@ -1405,15 +1385,9 @@ if (is_ios || is_mac) {
       }
 
       bundle_data("ios_framework_bundle") {
-        deps = [
+        deps = [ "../sdk:framework_objc" ]
-          "../sdk:framework_objc",
+        sources = [ "$root_build_dir/WebRTC.framework" ]
-        ]
+        outputs = [ "{{bundle_resources_dir}}/Frameworks/{{source_file_part}}" ]
-        sources = [
-          "$root_build_dir/WebRTC.framework",
-        ]
-        outputs = [
-          "{{bundle_resources_dir}}/Frameworks/{{source_file_part}}",
-        ]
       }
     }
 
@@ -1542,15 +1516,9 @@ if (is_ios || is_mac) {
       }
 
       bundle_data("mac_framework_bundle") {
-        deps = [
+        deps = [ "../sdk:mac_framework_objc" ]
-          "../sdk:mac_framework_objc",
+        sources = [ "$root_build_dir/WebRTC.framework" ]
-        ]
+        outputs = [ "{{bundle_contents_dir}}/Frameworks/{{source_file_part}}" ]
-        sources = [
-          "$root_build_dir/WebRTC.framework",
-        ]
-        outputs = [
-          "{{bundle_contents_dir}}/Frameworks/{{source_file_part}}",
-        ]
       }
     }
 

sdk/objc/native/src/audio/voice_processing_audio_unit.mm

@@ -10,8 +10,8 @@
 
 #import "voice_processing_audio_unit.h"
 
+#include "absl/base/macros.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/system/fallthrough.h"
 #include "system_wrappers/include/metrics.h"
 
 #import "base/RTCLogging.h"
@@ -446,12 +446,12 @@ void VoiceProcessingAudioUnit::DisposeAudioUnit() {
       case kStarted:
         Stop();
         // Fall through.
-        RTC_FALLTHROUGH();
+        ABSL_FALLTHROUGH_INTENDED;
       case kInitialized:
         Uninitialize();
         break;
       case kUninitialized:
-        RTC_FALLTHROUGH();
+        ABSL_FALLTHROUGH_INTENDED;
       case kInitRequired:
         break;
     }

video/BUILD.gn

@@ -110,7 +110,6 @@ rtc_library("video") {
     "../rtc_base/experiments:quality_scaling_experiment",
     "../rtc_base/experiments:rate_control_settings",
     "../rtc_base/synchronization:sequence_checker",
-    "../rtc_base/system:fallthrough",
     "../rtc_base/system:thread_registry",
     "../rtc_base/task_utils:repeating_task",
     "../rtc_base/task_utils:to_queued_task",
@@ -119,6 +118,7 @@ rtc_library("video") {
     "../system_wrappers:field_trial",
     "../system_wrappers:metrics",
     "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -230,11 +230,11 @@ rtc_library("video_stream_encoder_impl") {
     "../rtc_base/experiments:quality_scaling_experiment",
     "../rtc_base/experiments:rate_control_settings",
     "../rtc_base/synchronization:sequence_checker",
-    "../rtc_base/system:fallthrough",
     "../rtc_base/task_utils:repeating_task",
     "../system_wrappers",
     "../system_wrappers:field_trial",
     "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -242,9 +242,7 @@ rtc_library("video_stream_encoder_impl") {
 if (rtc_include_tests) {
   rtc_library("video_mocks") {
     testonly = true
-    sources = [
+    sources = [ "test/mock_video_stream_encoder.h" ]
-      "test/mock_video_stream_encoder.h",
-    ]
     deps = [
       "../api/video:video_stream_encoder",
       "../test:test_support",
@@ -330,9 +328,7 @@ if (rtc_include_tests) {
   rtc_library("video_full_stack_tests") {
     testonly = true
 
-    sources = [
+    sources = [ "full_stack_tests.cc" ]
-      "full_stack_tests.cc",
-    ]
     deps = [
       ":video_quality_test",
       "../api:simulated_network_api",
@@ -357,9 +353,7 @@ if (rtc_include_tests) {
   rtc_library("video_pc_full_stack_tests") {
     testonly = true
 
-    sources = [
+    sources = [ "pc_full_stack_tests.cc" ]
-      "pc_full_stack_tests.cc",
-    ]
     deps = [
       "../api:create_network_emulation_manager",
       "../api:create_peerconnection_quality_test_fixture",
@@ -408,31 +402,21 @@ if (rtc_include_tests) {
   if (is_mac) {
     mac_app_bundle("video_loopback") {
       testonly = true
-      sources = [
+      sources = [ "video_loopback_main.mm" ]
-        "video_loopback_main.mm",
-      ]
       info_plist = "../test/mac/Info.plist"
-      deps = [
+      deps = [ ":video_loopback_lib" ]
-        ":video_loopback_lib",
-      ]
     }
   } else {
     rtc_executable("video_loopback") {
       testonly = true
-      sources = [
+      sources = [ "video_loopback_main.cc" ]
-        "video_loopback_main.cc",
+      deps = [ ":video_loopback_lib" ]
-      ]
-      deps = [
-        ":video_loopback_lib",
-      ]
     }
   }
 
   rtc_executable("screenshare_loopback") {
     testonly = true
-    sources = [
+    sources = [ "screenshare_loopback.cc" ]
-      "screenshare_loopback.cc",
-    ]
 
     deps = [
       ":video_quality_test",
@@ -459,9 +443,7 @@ if (rtc_include_tests) {
 
   rtc_executable("sv_loopback") {
     testonly = true
-    sources = [
+    sources = [ "sv_loopback.cc" ]
-      "sv_loopback.cc",
-    ]
     deps = [
       ":video_quality_test",
       "../api:libjingle_peerconnection_api",
@@ -487,9 +469,7 @@ if (rtc_include_tests) {
 
   rtc_executable("video_replay") {
     testonly = true
-    sources = [
+    sources = [ "video_replay.cc" ]
-      "video_replay.cc",
-    ]
     deps = [
       "../api/rtc_event_log",
       "../api/task_queue:default_task_queue_factory",

video/buffered_frame_decryptor.cc

@@ -13,7 +13,6 @@
 #include <utility>
 
 #include "rtc_base/logging.h"
-#include "rtc_base/system/fallthrough.h"
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {

video/overuse_frame_detector_resource_adaptation_module.cc

@@ -17,13 +17,13 @@
 #include <utility>
 
 #include "absl/algorithm/container.h"
+#include "absl/base/macros.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/video/video_source_interface.h"
 #include "call/adaptation/video_source_restrictions.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/strings/string_builder.h"
-#include "rtc_base/system/fallthrough.h"
 #include "video/video_stream_encoder.h"
 
 namespace webrtc {
@@ -562,7 +562,7 @@ void OveruseFrameDetectorResourceAdaptationModule::AdaptUp(AdaptReason reason) {
         return;
       }
       // Scale up resolution.
-      RTC_FALLTHROUGH();
+      ABSL_FALLTHROUGH_INTENDED;
     }
     case DegradationPreference::MAINTAIN_FRAMERATE: {
       // Check if resolution should be increased based on bitrate and
@@ -678,7 +678,7 @@ bool OveruseFrameDetectorResourceAdaptationModule::AdaptDown(
         break;
       }
       // Scale down resolution.
-      RTC_FALLTHROUGH();
+      ABSL_FALLTHROUGH_INTENDED;
     }
     case DegradationPreference::MAINTAIN_FRAMERATE: {
       // Scale down resolution.

video/rtp_video_stream_receiver.cc

@@ -16,6 +16,7 @@
 #include <vector>
 
 #include "absl/algorithm/container.h"
+#include "absl/base/macros.h"
 #include "absl/memory/memory.h"
 #include "absl/types/optional.h"
 #include "media/base/media_constants.h"
@@ -43,7 +44,6 @@
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
-#include "rtc_base/system/fallthrough.h"
 #include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 #include "video/receive_statistics_proxy.h"
@@ -467,7 +467,7 @@ void RtpVideoStreamReceiver::OnReceivedPayloadData(
       case video_coding::H264SpsPpsTracker::kRequestKeyframe:
         rtcp_feedback_buffer_.RequestKeyFrame();
         rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
-        RTC_FALLTHROUGH();
+        ABSL_FALLTHROUGH_INTENDED;
       case video_coding::H264SpsPpsTracker::kDrop:
         return;
       case video_coding::H264SpsPpsTracker::kInsert:

video/video_stream_encoder.cc

@@ -34,7 +34,6 @@
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
-#include "rtc_base/system/fallthrough.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"