Build WebRTC with data channel only.

webrtc/media/BUILD.gn

Index: webrtc/media/BUILD.gn
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index cc775984398c50f7ab14e399d9f024923def7a90..2cfe6f905ad8bf9c777444f6d647ee5d8dad4bba 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -11,6 +11,7 @@ import("../webrtc.gni")
 
 group("media") {
   public_deps = [
+    ":rtc_audio_video",
     ":rtc_media",
     ":rtc_media_base",
   ]
@@ -32,6 +33,29 @@ config("rtc_media_warnings_config") {
   }
 }
 
+rtc_static_library("rtc_media_audio_base") {
+  deps = [
+    "../api/audio_codecs:audio_codecs_api",
+  ]
+}
+
+rtc_static_library("rtc_media_video_base_nullimpl") {
+  sources = [
+    "base/codec_video_nullimpl.cc",
+  ]
+}
+
+rtc_static_library("rtc_media_video_base") {
+  sources = [
+    "base/codec_video.cc",
+  ]
+
+  deps = [
+    "../api:video_frame_api",
+    "../common_video:common_video",
+  ]
+}
+
 rtc_static_library("rtc_media_base") {
   # TODO(kjellander): Remove (bugs.webrtc.org/6828)
   # Enabling GN check triggers cyclic dependency error:
@@ -98,13 +122,9 @@ rtc_static_library("rtc_media_base") {
   deps += [
     "..:webrtc_common",
     "../api:libjingle_peerconnection_api",
-    "../api:video_frame_api",
-    "../api/audio_codecs:audio_codecs_api",
-    "../api/audio_codecs:builtin_audio_encoder_factory",
     "../base:rtc_base",
     "../base:rtc_base_approved",
     "../call:call_interfaces",
-    "../common_video:common_video",
     "../p2p",
   ]
 
@@ -113,7 +133,7 @@ rtc_static_library("rtc_media_base") {
   }
 }
 
-rtc_static_library("rtc_media") {
+rtc_static_library("rtc_audio_video") {
   # TODO(kjellander): Remove (bugs.webrtc.org/6828)
   # Enabling GN check triggers cyclic dependency error:
   #   //webrtc/media:media ->
@@ -156,16 +176,8 @@ rtc_static_library("rtc_media") {
     "engine/webrtcvoe.h",
     "engine/webrtcvoiceengine.cc",
     "engine/webrtcvoiceengine.h",
-    "sctp/sctptransportinternal.h",
   ]
 
-  if (rtc_enable_sctp) {
-    sources += [
-      "sctp/sctptransport.cc",
-      "sctp/sctptransport.h",
-    ]
-  }
-
   configs += [ ":rtc_media_warnings_config" ]
 
   if (!build_with_chromium && is_clang) {
@@ -204,15 +216,6 @@ rtc_static_library("rtc_media") {
     include_dirs += [ "$rtc_libyuv_dir/include" ]
   }
 
-  if (rtc_enable_sctp && rtc_build_usrsctp) {
-    include_dirs += [
-      # TODO(jiayl): move this into the public_configs of
-      # //third_party/usrsctp/BUILD.gn.
-      "//third_party/usrsctp/usrsctplib",
-    ]
-    deps += [ "//third_party/usrsctp" ]
-  }
-
   public_configs = []
   if (build_with_chromium) {
     deps += [ "../modules/video_capture:video_capture" ]
@@ -221,7 +224,9 @@ rtc_static_library("rtc_media") {
     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
   }
   deps += [
+    ":rtc_media_audio_base",
     ":rtc_media_base",
+    ":rtc_media_video_base",
     "..:webrtc_common",
     "../api:call_api",
     "../api:transport_api",
@@ -249,6 +254,61 @@ rtc_static_library("rtc_media") {
   ]
 }
 
+rtc_static_library("rtc_media") {
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Enabling GN check triggers cyclic dependency error:
+  #   //webrtc/media:media ->
+  #   //webrtc/media:rtc_media ->
+  #   //webrtc/pc:rtc_pc ->
+  #   //webrtc/media:media
+  check_includes = false
+  defines = []
+  deps = []
+
+  if (rtc_enable_sctp) {
+    sources = [
+      "sctp/sctptransport.cc",
+      "sctp/sctptransport.h",
+      "sctp/sctptransportinternal.h",
+    ]
+  }
+
+  configs += [ ":rtc_media_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" ]
+  }
+
+  if (is_win) {
+    cflags = [
+      "/wd4245",  # conversion from "int" to "size_t", signed/unsigned mismatch.
+      "/wd4267",  # conversion from "size_t" to "int", possible loss of data.
+      "/wd4389",  # signed/unsigned mismatch.
+    ]
+  }
+
+  if (rtc_enable_sctp && rtc_build_usrsctp) {
+    include_dirs = [
+      # TODO(jiayl): move this into the public_configs of
+      # //third_party/usrsctp/BUILD.gn.
+      "//third_party/usrsctp/usrsctplib",
+    ]
+    deps += [ "//third_party/usrsctp" ]
+  }
+
+  deps += [
+    ":rtc_media_base",
+    "..:webrtc_common",
+    "../api:call_api",
+    "../api:transport_api",
+    "../base:rtc_base",
+    "../base:rtc_base_approved",
+    "../p2p:rtc_p2p",
+    "../system_wrappers",
+  ]
+}
+
 if (rtc_include_tests) {
   config("rtc_unittest_main_config") {
     # GN orders flags on a target before flags from configs. The default config