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 4e4d67a9376dc3fb52321863467f283f65b0cd17..97a77d7c09474bbb803cdf09a0b97dc007939ea3 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -32,7 +32,7 @@ config("rtc_media_warnings_config") {
   }
 }
 
-rtc_static_library("rtc_media_base") {
+rtc_source_set("rtc_media_base_common") {
   # TODO(kjellander): Remove (bugs.webrtc.org/6828)
   # Enabling GN check triggers cyclic dependency error:
   #   //webrtc/media:rtc_media_base ->
@@ -40,15 +40,11 @@ rtc_static_library("rtc_media_base") {
   #   //webrtc/media:media ->
   #   //webrtc/media:rtc_media_base
   check_includes = false
-  defines = []
-  libs = []
   deps = []
   sources = [
     "base/adaptedvideotracksource.cc",
     "base/adaptedvideotracksource.h",
     "base/audiosource.h",
-    "base/codec.cc",
-    "base/codec.h",
     "base/cryptoparams.h",
     "base/device.h",
     "base/mediachannel.h",
@@ -98,13 +94,10 @@ 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",
   ]
 
@@ -113,6 +106,119 @@ rtc_static_library("rtc_media_base") {
   }
 }
 
+rtc_static_library("rtc_media_base_datachannel_only") {
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Enabling GN check triggers cyclic dependency error:
+  #   //webrtc/media:rtc_media_base ->
+  #   //webrtc/pc:rtc_pc ->
+  #   //webrtc/media:media ->
+  #   //webrtc/media:rtc_media_base
+  check_includes = false
+  defines = []
+  sources = [
+    "base/codec.cc",
+    "base/codec.h",
+  ]
+
+  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" ]
+  }
+
+  public_deps = [
+    ":rtc_media_base_common",
+  ]
+}
+
+rtc_static_library("rtc_media_base") {
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Enabling GN check triggers cyclic dependency error:
+  #   //webrtc/media:rtc_media_base ->
+  #   //webrtc/pc:rtc_pc ->
+  #   //webrtc/media:media ->
+  #   //webrtc/media:rtc_media_base
+  check_includes = false
+  defines = [
+    "HAVE_WEBRTC_VOICE",
+    "HAVE_WEBRTC_VIDEO",
+  ]
+  libs = []
+  deps = []
+  sources = [
+    "base/codec.cc",
+    "base/codec.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" ]
+  }
+
+  deps += [
+    "../api:video_frame_api",
+    "../api/audio_codecs:builtin_audio_encoder_factory",
+    "../common_video:common_video",
+  ]
+
+  public_deps = [
+    ":rtc_media_base_common",
+  ]
+}
+
+rtc_static_library("rtc_media_datachannel_only") {
+  # 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 = []
+  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_datachannel_only",
+    "..:webrtc_common",
+    "../api:call_api",
+    "../api:transport_api",
+    "../base:rtc_base",
+    "../base:rtc_base_approved",
+    "../p2p:rtc_p2p",
+    "../system_wrappers",
+  ]
+}
+
 rtc_static_library("rtc_media") {
   # TODO(kjellander): Remove (bugs.webrtc.org/6828)
   # Enabling GN check triggers cyclic dependency error: