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..6702b11ba6bb93d576a47d6b6e89ffe8b485b6c3 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -82,6 +82,11 @@ rtc_static_library("rent_a_codec") {
            ":isac_fix_c",
            ":neteq_decoder_enum",
          ] + audio_codec_deps
+
+  if (rtc_enable_protobuf) {
+    deps += [ "../../base:protobuf_utils" ]
+  }

[kwiberg-webrtc, 2017/03/22 10:03:23]

Why would you ever want a conditional dep on protobuf_utils? I thought the idea
was that protobuf_utils / protobuf_utils.h would always be there, but would be
empty if we don't have protobuf support.

In the .h and .cc files, you #include protobuf_utils.h unconditionally. That's
what you should do if you follow my advice, but if you want to do the
conditional deps like you do here, you need to protect those #includes with #if
WEBRTC_PROTOBUF_ENABLED. Which would be ugly... but if you don't do it, gn check
ought to complain if you build with rtc_enable_protobuf=false. (Did you try
that?)

[mbonadei, 2017/03/23 16:13:58]

Done and thanks for spotting this inconsistency. My bad, I was going in two
opposite directions.

Yes, I tried 'gn check' (both with rtc_enable_protobuf = true and
rtc_enable_protobuf = false) but it was somehow happy with all the conditions on
rtc_enable_protobuf. I guess that if the #include is not part of a defined
target it doesn't complain.

+
   defines = audio_codec_defines
 }
 
@@ -837,6 +842,10 @@ 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 +937,7 @@ rtc_static_library("audio_network_adaptor") {
     deps += [
       ":ana_config_proto",
       ":ana_debug_dump_proto",
+      "../../base:protobuf_utils",
     ]
     defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
   }
@@ -1190,6 +1200,11 @@ if (rtc_include_tests) {
       "../../system_wrappers:system_wrappers",
       "../../test:test_support",
     ]
+
+    if (rtc_enable_protobuf) {
+      deps += [ "../../base:protobuf_utils" ]
+    }
+
     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" ]
@@ -1329,6 +1344,10 @@ if (rtc_include_tests) {
       "//testing/gtest",
     ]
 
+    if (rtc_enable_protobuf) {
+      deps += [ "../../base:protobuf_utils" ]
+    }
+
     data = audio_decoder_unittests_resources
 
     if (is_android) {
@@ -2102,6 +2121,7 @@ if (rtc_include_tests) {
       deps += [
         ":ana_config_proto",
         ":neteq_unittest_proto",
+        "../../base:protobuf_utils",
       ]
     }