Revert of Loosening the coupling between WebRTC and //third_party/protobuf

webrtc/modules/audio_coding/BUILD.gn

 # Copyright (c) 2014 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("../../webrtc.gni")
 import("audio_coding.gni")
@@ skipping 56 matching lines @@
 rtc_static_library("rent_a_codec") {
   sources = [
     "acm2/acm_codec_database.cc",
     "acm2/acm_codec_database.h",
     "acm2/rent_a_codec.cc",
     "acm2/rent_a_codec.h",
   ]
   deps = [
            "../../api/audio_codecs:audio_codecs_api",
            "../..:webrtc_common",
-           "../../base:protobuf_utils",
            "../../base:rtc_base_approved",
            "../../system_wrappers",
            ":audio_coding_module_typedefs",
            ":audio_encoder_interface",
            ":isac_common",
            ":isac_fix_c",
            ":neteq_decoder_enum",
          ] + audio_codec_deps
-
   defines = audio_codec_defines
 }
 
 config("audio_coding_config") {
   include_dirs = [
     "include",
     "../include",
   ]
 }
 
@@ skipping 726 matching lines @@
     "codecs/opus/audio_decoder_opus.h",
     "codecs/opus/audio_encoder_opus.cc",
     "codecs/opus/audio_encoder_opus.h",
   ]
 
   deps = [
     ":audio_encoder_interface",
     ":audio_network_adaptor",
     "../..:webrtc_common",
     "../../api/audio_codecs:audio_codecs_api",
-    "../../base:protobuf_utils",
     "../../base:rtc_base_approved",
     "../../base:rtc_numerics",
     "../../common_audio",
     "../../system_wrappers",
   ]
   public_deps = [
     ":webrtc_opus_c",
   ]
 
   defines = audio_codec_defines
@@ skipping 73 matching lines @@
     "audio_network_adaptor/fec_controller_rplr_based.cc",
     "audio_network_adaptor/fec_controller_rplr_based.h",
     "audio_network_adaptor/frame_length_controller.cc",
     "audio_network_adaptor/frame_length_controller.h",
     "audio_network_adaptor/include/audio_network_adaptor.h",
     "audio_network_adaptor/util/threshold_curve.h",
   ]
 
   deps = [
     "../..:webrtc_common",
-    "../../base:protobuf_utils",
     "../../base:rtc_base_approved",
     "../../common_audio",
     "../../logging:rtc_event_log_api",
     "../../system_wrappers",
   ]
 
   if (rtc_enable_protobuf) {
     deps += [
       ":ana_config_proto",
       ":ana_debug_dump_proto",
@@ skipping 247 matching lines @@
     testonly = true
     sources = [
       "codecs/opus/opus_complexity_unittest.cc",
       "neteq/test/neteq_performance_unittest.cc",
     ]
     deps = [
       ":neteq_test_support",
       ":neteq_unittest_tools",
       ":webrtc_opus",
       "../..:webrtc_common",
-      "../../base:protobuf_utils",
       "../../base:rtc_base_approved",
       "../../system_wrappers:system_wrappers",
       "../../test:test_support",
     ]
-
     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" ]
     }
   }
 
   rtc_source_set("acm_receive_test") {
     testonly = true
     sources = [
       "acm2/acm_receive_test.cc",
@@ skipping 114 matching lines @@
     defines = neteq_defines
 
     deps += audio_coding_deps
     deps += [
       ":ilbc",
       ":isac",
       ":isac_fix",
       ":neteq",
       ":neteq_unittest_tools",
       "../../api/audio_codecs:audio_codecs_api",
-      "../../base:protobuf_utils",
       "../../common_audio",
       "../../test:test_main",
       "//testing/gtest",
     ]
 
     data = audio_decoder_unittests_resources
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
       shard_timeout = 900
@@ skipping 736 matching lines @@
       ":neteq",
       ":neteq_test_support",
       ":neteq_unittest_tools",
       ":pcm16b",
       ":red",
       ":rent_a_codec",
       ":webrtc_opus",
       "../..:webrtc_common",
       "../../api/audio_codecs:audio_codecs_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
-      "../../base:protobuf_utils",
       "../../base:rtc_base",
       "../../base:rtc_base_approved",
       "../../base:rtc_base_tests_utils",
       "../../common_audio",
       "../../system_wrappers:system_wrappers",
       "../../test:field_trial",
       "../../test:rtp_test_utils",
       "../../test:test_common",
       "../../test:test_support",
       "//testing/gmock",
@@ skipping 37 matching lines @@
 # webrtc/api/audio_codecs:builtin_audio_decoder_factory instead.
 # TODO(kwiberg): Remove this.
 rtc_source_set("builtin_audio_decoder_factory") {
   sources = [
     "codecs/builtin_audio_decoder_factory.h",
   ]
   deps = [
     "../../api/audio_codecs:builtin_audio_decoder_factory",
   ]
 }