Build WebRTC with data channel only.

webrtc/media/BUILD.gn

 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("//build/config/linux/pkg_config.gni")
 import("../webrtc.gni")
@@ skipping 14 matching lines @@
 
 config("rtc_media_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
   # cannot be on the target directly.
   if (!is_win) {
     cflags = [ "-Wno-deprecated-declarations" ]
   }
 }
 
-rtc_static_library("rtc_media_base") {
+rtc_source_set("rtc_media_base_audio") {
+  sources = [
+    "base/audiosource.h",
+  ]
+  deps = [
+    "../api/audio_codecs:audio_codecs_api",
+  ]
+}
+
+rtc_source_set("rtc_media_base_video") {
+  deps = [
+    "../api:video_frame_api",
+    "../common_video:common_video",
+  ]
+}
+
+# This target is used to build WebRTC without audio and video support but it
+# contains more than just datachannel related classes.
+# TODO(zhihuang): Split this target further into a target containing only
+# datachannel related classes and a target containing common classes for media
+# base.
+rtc_source_set("rtc_media_base_data") {
   # 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
+  #   //webrtc/media:rtc_media_base_data ->
+  #   //webrtc/pc:rtc_pc_base ->
+  #   //webrtc/media:rtc_data ->
+  #   //webrtc/media:rtc_media_base_data
   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/h264_profile_level_id.cc",
+    "base/h264_profile_level_id.h",
     "base/mediachannel.h",
     "base/mediaconstants.cc",
     "base/mediaconstants.h",
     "base/mediaengine.cc",
     "base/mediaengine.h",
     "base/rtpdataengine.cc",
     "base/rtpdataengine.h",
     "base/rtputils.cc",
     "base/rtputils.h",
     "base/streamparams.cc",
@@ skipping 27 matching lines @@
       "$rtc_libyuv_dir",
     ]
   } else {
     # Need to add a directory normally exported by libyuv.
     include_dirs += [ "$rtc_libyuv_dir/include" ]
   }
 
   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",
   ]
 
   if (is_nacl) {
     deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
   }
 }
 
-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
+rtc_source_set("rtc_media_base") {
+  public_deps = [
+    ":rtc_media_base_audio",
+    ":rtc_media_base_data",
+    ":rtc_media_base_video",
+  ]
+}
+
+rtc_static_library("rtc_audio_video") {
   defines = []
   libs = []
   deps = []
   sources = [
     "engine/adm_helpers.cc",
     "engine/adm_helpers.h",
     "engine/apm_helpers.cc",
     "engine/apm_helpers.h",
     "engine/internaldecoderfactory.cc",
     "engine/internaldecoderfactory.h",
@@ skipping 15 matching lines @@
     "engine/webrtcvideocapturer.h",
     "engine/webrtcvideocapturerfactory.cc",
     "engine/webrtcvideocapturerfactory.h",
     "engine/webrtcvideodecoderfactory.h",
     "engine/webrtcvideoencoderfactory.h",
     "engine/webrtcvideoengine2.cc",
     "engine/webrtcvideoengine2.h",
     "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) {
     # 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.
@@ skipping 18 matching lines @@
   if (rtc_build_libyuv) {
     deps += [ "$rtc_libyuv_dir" ]
     public_deps = [
       "$rtc_libyuv_dir",
     ]
   } else {
     # Need to add a directory normally exported by libyuv.
     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" ]
   } else {
     public_configs += [ ":rtc_media_defines_config" ]
     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
   }
   if (rtc_enable_protobuf) {
     deps += [ "../modules/audio_processing/aec_dump:aec_dump_impl" ]
   } else {
     deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
   }
   deps += [
-    ":rtc_media_base",
+    ":rtc_media_base_audio",
+    ":rtc_media_base_data",
+    ":rtc_media_base_video",
+    "..:video_stream_api",
     "..:webrtc_common",
     "../api:call_api",
+    "../api:libjingle_peerconnection_api",
     "../api:transport_api",
     "../api:video_frame_api",
     "../api/audio_codecs:audio_codecs_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",
+    "../base:rtc_task_queue",
     "../call",
     "../common_video:common_video",
     "../modules/audio_coding:rent_a_codec",
     "../modules/audio_device:audio_device",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/audio_processing:audio_processing",
     "../modules/audio_processing/aec_dump",
     "../modules/video_capture:video_capture_module",
     "../modules/video_coding",
     "../modules/video_coding:webrtc_h264",
     "../modules/video_coding:webrtc_vp8",
     "../modules/video_coding:webrtc_vp9",
     "../p2p:rtc_p2p",
+    "../pc:rtc_pc_base",
     "../system_wrappers",
     "../video",
     "../voice_engine",
   ]
 }
 
+rtc_static_library("rtc_data") {
+  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_data",
+    "..:webrtc_common",
+    "../api:call_api",
+    "../api:transport_api",
+    "../base:rtc_base",
+    "../base:rtc_base_approved",
+    "../p2p:rtc_p2p",
+    "../system_wrappers",
+  ]
+}
+
+rtc_source_set("rtc_media") {

[kjellander_webrtc, 2017/06/01 05:34:30]

Can we keep rtc_static_library ?
See https://codereview.webrtc.org/2361623004 and
https://bugs.chromium.org/p/chromium/issues/detail?id=630755#c1 for context. TBH
I don't remember exactly the criteria where we should have it or not though.

[Zhi Huang, 2017/06/02 05:16:43]

It seems that the rtc_static_library requires at least one source file to
compile otherwise it will fail (on win and mac at least).

[kjellander_webrtc, 2017/06/02 06:46:18]

I remember hitting problems like this in the past but it's actually just a lot
of warnings on Mac - the linking succeeds and exits with code 0.
I don't remember exactly how it looks on Windows but since the 'webrtc' target
https://cs.chromium.org/chromium/src/third_party/webrtc/BUILD.gn?rcl=7d76f48e...
is able to build (call.h doesn't change it and is being dropped in
https://codereview.webrtc.org/2916513002/ without build errors).

But if you're sure you're getting an error if changing this, I'm there's nothing
to do about it.

+  public_deps = [
+    ":rtc_audio_video",
+    ":rtc_data",
+  ]
+}
+
 if (rtc_include_tests) {
   config("rtc_unittest_main_config") {
     # GN orders flags on a target before flags from configs. The default config
     # adds -Wall, and this flag have to be after -Wall -- so they need to
     # come from a config and can"t be on the target directly.
     if (is_clang && is_ios) {
       cflags = [ "-Wno-unused-variable" ]
     }
   }
 
@@ skipping 195 matching lines @@
       "../modules/video_coding:video_coding_utility",
       "../modules/video_coding:webrtc_vp8",
       "../p2p:p2p_test_utils",
       "../system_wrappers:metrics_default",
       "../test:audio_codec_mocks",
       "../test:test_support",
       "../voice_engine:voice_engine",
     ]
   }
 }