Loosening the coupling between WebRTC and //third_party/protobuf

webrtc/modules/audio_coding/BUILD.gn

Index: webrtc/modules/audio_coding/BUILD.gn
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 3a2d20a800e8326fe4b6077992474c7213ffd31d..96672909613cf542f4f091149526b0530c1865da 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -837,6 +837,10 @@ rtc_static_library("webrtc_opus") {
     ":webrtc_opus_c",
   ]
 
+  if (rtc_enable_protobuf) {
+    deps += [ "../../base:protobuf_utils" ]
+  }
+
   defines = audio_codec_defines
   if (rtc_opus_variable_complexity) {
     defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
@@ -928,6 +932,7 @@ rtc_static_library("audio_network_adaptor") {
     deps += [
       ":ana_config_proto",
       ":ana_debug_dump_proto",
+      "../../base:protobuf_utils",
     ]
     defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
   }
@@ -2102,6 +2107,7 @@ if (rtc_include_tests) {
       deps += [
         ":ana_config_proto",
         ":neteq_unittest_proto",
+        "../../base:protobuf_utils",
       ]
     }