Loosening the coupling between WebRTC and //third_party/protobuf

webrtc/modules/audio_processing/BUILD.gn

Index: webrtc/modules/audio_processing/BUILD.gn
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 84f25b5e8ed7cfae29013af1fb8fba48e5805ef3..c9bc01bdaadaea1268577e3c92695f7a7599ce85 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -250,7 +250,10 @@ rtc_static_library("audio_processing") {
 
   if (rtc_enable_protobuf) {
     defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
-    deps += [ ":audioproc_debug_proto" ]
+    deps += [
+      ":audioproc_debug_proto",
+      "../../base:protobuf_utils",
+    ]
   }
 
   if (rtc_enable_intelligibility_enhancer) {
@@ -563,6 +566,7 @@ if (rtc_include_tests) {
         ":audioproc_debug_proto",
         ":audioproc_protobuf_utils",
         ":audioproc_unittest_proto",
+        "../../base:protobuf_utils",
       ]
       sources += [
         "aec3/adaptive_fir_filter_unittest.cc",
@@ -656,6 +660,11 @@ if (rtc_include_tests) {
       ":audioproc_test_utils",
       "//testing/gtest",
     ]
+
+    if (rtc_enable_protobuf) {
+      deps += [ "../../base:protobuf_utils" ]
+    }
+
     if (rtc_enable_intelligibility_enhancer) {
       defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
     } else {
@@ -815,6 +824,7 @@ if (rtc_include_tests) {
       deps = [
         ":audioproc_debug_proto",
         "../..:webrtc_common",
+        "../../base:protobuf_utils",
         "../../base:rtc_base_approved",
       ]
     }