Reland of Try to fix the binary size increase issue on Chromium.

webrtc/media/BUILD.gn

 # Copyright (c) 2016 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.
 
 import("//build/config/linux/pkg_config.gni")
 import("../webrtc.gni")
@@ skipping 14 matching lines @@
 
 config("rtc_media_warnings_config") {
   # GN orders flags on a target before flags from configs. The default config
   # adds these flags so to cancel them out they need to come from a config and
   # cannot be on the target directly.
   if (!is_win) {
     cflags = [ "-Wno-deprecated-declarations" ]
   }
 }
 
+rtc_source_set("rtc_h264_profile_id") {
+  sources = [
+    "base/h264_profile_level_id.cc",
+    "base/h264_profile_level_id.h",
+  ]
+
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+
+  deps = [
+    "..:webrtc_common",
+    "../base:rtc_base",
+    "../base:rtc_base_approved",
+  ]
+}
+
 rtc_source_set("rtc_media_base") {
   # TODO(kjellander): Remove (bugs.webrtc.org/6828)
   # Enabling GN check triggers cyclic dependency error:
   #   //webrtc/media:rtc_media_base ->
   #   //webrtc/pc:rtc_pc_base ->
   #   //webrtc/media:rtc_data ->
   #   //webrtc/media:rtc_media_base
   check_includes = false
   defines = []
   libs = []
   deps = []
   sources = [
     "base/adaptedvideotracksource.cc",
     "base/adaptedvideotracksource.h",
     "base/audiosource.h",
     "base/codec.cc",
     "base/codec.h",
     "base/cryptoparams.h",
     "base/device.h",
-    "base/h264_profile_level_id.cc",
-    "base/h264_profile_level_id.h",
     "base/mediachannel.h",
     "base/mediaconstants.cc",
     "base/mediaconstants.h",
     "base/mediaengine.cc",
     "base/mediaengine.h",
     "base/rtpdataengine.cc",
     "base/rtpdataengine.h",
     "base/rtputils.cc",
     "base/rtputils.h",
     "base/streamparams.cc",
@@ skipping 23 matching lines @@
     deps += [ "$rtc_libyuv_dir" ]
     public_deps = [
       "$rtc_libyuv_dir",
     ]
   } else {
     # Need to add a directory normally exported by libyuv.
     include_dirs += [ "$rtc_libyuv_dir/include" ]
   }
 
   deps += [
+    ":rtc_h264_profile_id",
     "..:webrtc_common",
     "../api:libjingle_peerconnection_api",
     "../base:rtc_base",
     "../base:rtc_base_approved",
     "../p2p",
   ]
 
   if (is_nacl) {
     deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
   }
@@ skipping 385 matching lines @@
       "../modules/video_coding:video_coding_utility",
       "../modules/video_coding:webrtc_vp8",
       "../p2p:p2p_test_utils",
       "../system_wrappers:metrics_default",
       "../test:audio_codec_mocks",
       "../test:test_support",
       "../voice_engine:voice_engine",
     ]
   }
 }