Build WebRTC with data channel only.

webrtc/pc/BUILD.gn

Index: webrtc/pc/BUILD.gn
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index 10256ce2f407cc3b76e5ee3c28dfbde8188536be..55013ee389da02d51968b36b0dd27076f7f5bf88 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -25,7 +25,7 @@ config("rtc_pc_config") {
   }
 }
 
-rtc_static_library("rtc_pc") {
+rtc_static_library("rtc_pc_base") {
   defines = []
   sources = [
     "audiomonitor.cc",
@@ -59,8 +59,9 @@ rtc_static_library("rtc_pc") {
     "../api:libjingle_peerconnection_api",
     "../api:ortc_api",
     "../base:rtc_base",
-    "../common_video:common_video",
-    "../media",
+    "../base:rtc_task_queue",
+    "../media:rtc_data",
+    "../media:rtc_media_base",
     "../p2p:rtc_p2p",
   ]
 
@@ -76,6 +77,16 @@ rtc_static_library("rtc_pc") {
   }
 }
 
+rtc_source_set("rtc_pc") {
+  public_deps = [
+    ":rtc_pc_base",
+  ]
+
+  deps = [
+    "../media:rtc_audio_video",
+  ]
+}
+
 config("libjingle_peerconnection_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
@@ -85,7 +96,7 @@ config("libjingle_peerconnection_warnings_config") {
   }
 }
 
-rtc_static_library("libjingle_peerconnection") {
+rtc_static_library("peerconnection") {
   cflags = []
   sources = [
     "audiotrack.cc",
@@ -146,19 +157,17 @@ rtc_static_library("libjingle_peerconnection") {
   }
 
   deps = [
-    ":rtc_pc",
+    ":rtc_pc_base",
     "..:webrtc_common",
     "../api:call_api",
     "../api:rtc_stats_api",
-    "../api/audio_codecs:builtin_audio_decoder_factory",
-    "../api/audio_codecs:builtin_audio_encoder_factory",
     "../api/video_codecs:video_codecs_api",
     "../base:rtc_base",
     "../base:rtc_base_approved",
-    "../call",
+    "../call:call_interfaces",
     "../logging:rtc_event_log_api",
-    "../media",
-    "../modules/audio_device:audio_device",
+    "../media:rtc_data",
+    "../media:rtc_media_base",
     "../p2p:rtc_p2p",
     "../stats",
     "../system_wrappers:system_wrappers",
@@ -167,6 +176,48 @@ rtc_static_library("libjingle_peerconnection") {
   public_deps = [
     "../api:libjingle_peerconnection_api",
   ]
+}
+
+# This target implements CreatePeerConnectionFactory methods that will create a
+# PeerConnection will full functionality (audio, video and data). Applications
+# that wish to reduce their binary size by ommitting functionality they don't
+# need should use CreateModularCreatePeerConnectionFactory instead, using the
+# "peerconnection" build target and other targets specific to their
+# requrements. See comment in peerconnectionfactoryinterface.h.

[Taylor Brandstetter, 2017/06/15 00:35:02]

Just noticed my own typo, "requrements"

+rtc_static_library("create_pc_factory") {

[Taylor Brandstetter, 2017/06/15 07:31:04]

If you change this to rtc_source_set, the iOS simulator link error goes away. I
have no idea what's going on, but it must be something to do with linker
optimizations. It also goes away if GetSources stops being virtual, or if
"kDefaultStartBitrateBps" moves back to call.cc; both of those changes
re-introduce a dependency between "rtc_pc" and "rtc_audio_video".

This blog post may shed some light:
https://groups.google.com/a/chromium.org/forum/#!topic/gn-dev/Jn4TbhOtB9k

I'm still not sure what precisely is going on, though. Maybe kjellander@ can
explain.

[kjellander_webrtc, 2017/06/15 09:08:50]

I'm sorry I don't have a better explanation. You should probably loop in brettw@
if you really want to know the details.

+  sources = [
+    "createpeerconnectionfactory.cc",
+  ]
+
+  deps = [
+    "../api:audio_mixer_api",
+    "../api:libjingle_peerconnection_api",
+    "../api/audio_codecs:audio_codecs_api",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
+    "../api/audio_codecs:builtin_audio_encoder_factory",
+    "../base:rtc_base",
+    "../base:rtc_base_approved",
+    "../call",
+    "../call:call_interfaces",
+    "../logging:rtc_event_log_api",
+    "../media:rtc_audio_video",
+    "../modules/audio_device:audio_device",
+  ]
+
+  configs += [ ":libjingle_peerconnection_warnings_config" ]
+
+  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("libjingle_peerconnection") {
+  public_deps = [
+    ":create_pc_factory",
+    ":peerconnection",
+    "../api:libjingle_peerconnection_api",
+  ]
 
   if (rtc_use_quic) {
     sources += [
@@ -271,6 +322,8 @@ if (rtc_include_tests) {
       "../base:rtc_base",
       "../base:rtc_base_approved",
       "../base:rtc_base_tests_utils",
+      "../call:call_interfaces",
+      "../logging:rtc_event_log_api",
       "../media:rtc_media",
       "../media:rtc_media_tests_utils",
       "../modules/audio_device:audio_device",