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 81 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" ]
   }
+
+  if (rtc_enable_media) {
+    deps += [
+      "../api:video_frame_api",
+      "../api/audio_codecs:builtin_audio_encoder_factory",
+      "../common_video:common_video",
+      "../modules/audio_coding:builtin_audio_encoder_factory",
+    ]
+  }
 }
 
 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 = []
   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",
-    "engine/internalencoderfactory.cc",
-    "engine/internalencoderfactory.h",
-    "engine/nullwebrtcvideoengine.h",
-    "engine/payload_type_mapper.cc",
-    "engine/payload_type_mapper.h",
-    "engine/simulcast.cc",
-    "engine/simulcast.h",
-    "engine/videodecodersoftwarefallbackwrapper.cc",
-    "engine/videodecodersoftwarefallbackwrapper.h",
-    "engine/videoencodersoftwarefallbackwrapper.cc",
-    "engine/videoencodersoftwarefallbackwrapper.h",
-    "engine/webrtccommon.h",
-    "engine/webrtcmediaengine.cc",
-    "engine/webrtcmediaengine.h",
-    "engine/webrtcvideocapturer.cc",
-    "engine/webrtcvideocapturer.h",
-    "engine/webrtcvideocapturerfactory.cc",
-    "engine/webrtcvideocapturerfactory.h",
     "engine/webrtcvideodecoderfactory.h",
     "engine/webrtcvideoencoderfactory.cc",
     "engine/webrtcvideoencoderfactory.h",

[Taylor Brandstetter, 2017/05/03 22:50:57]

What would it take to move these files into the "if (rtc_enable_media)"? It
would be nice if we could do that, since then we'd have one build target that's
only SCTP stuff.

[Zhi Huang, 2017/05/04 01:08:02]

The dependency chain is:
peerconnection_integrationtests -> fakevideoencoerfactory ->
webrtcvideoencoderfactroy.

They will just affect the tests. We'll be fine assuming the tests are never
build in the data-only mode. Do you think we can assume that?

[pthatcher1, 2017/05/05 01:45:12]

If they do, it will be really easy to know and fix it (the build would be
broken), so I think it would be OK to make that assumption.

-    "engine/webrtcvideoengine2.cc",
-    "engine/webrtcvideoengine2.h",
-    "engine/webrtcvideoframe.h",
-    "engine/webrtcvoe.h",
-    "engine/webrtcvoiceengine.cc",
-    "engine/webrtcvoiceengine.h",
     "sctp/sctptransportinternal.h",
   ]
 
   if (rtc_enable_sctp) {
     sources += [
       "sctp/sctptransport.cc",
       "sctp/sctptransport.h",
     ]
   }
 
@@ skipping 49 matching lines @@
     deps += [ "../modules/video_capture:video_capture" ]
   } else {
     public_configs += [ ":rtc_media_defines_config" ]
     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
   }
   deps += [
     ":rtc_media_base",
     "..:webrtc_common",
     "../api:call_api",
     "../api:transport_api",
-    "../api:video_frame_api",
     "../api/audio_codecs:audio_codecs_api",
-    "../api/audio_codecs:builtin_audio_decoder_factory",
     "../api/video_codecs:video_codecs_api",
     "../base:rtc_base",
     "../base:rtc_base_approved",
-    "../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/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",
     "../system_wrappers",
-    "../video",
-    "../voice_engine",
   ]
+
+  if (rtc_enable_media) {
+    sources += [
+      "engine/adm_helpers.cc",
+      "engine/adm_helpers.h",
+      "engine/apm_helpers.cc",
+      "engine/apm_helpers.h",
+      "engine/internaldecoderfactory.cc",
+      "engine/internaldecoderfactory.h",
+      "engine/internalencoderfactory.cc",
+      "engine/internalencoderfactory.h",
+      "engine/nullwebrtcvideoengine.h",
+      "engine/payload_type_mapper.cc",
+      "engine/payload_type_mapper.h",
+      "engine/simulcast.cc",
+      "engine/simulcast.h",
+      "engine/videodecodersoftwarefallbackwrapper.cc",
+      "engine/videodecodersoftwarefallbackwrapper.h",
+      "engine/videoencodersoftwarefallbackwrapper.cc",
+      "engine/videoencodersoftwarefallbackwrapper.h",
+      "engine/webrtccommon.h",
+      "engine/webrtcmediaengine.cc",
+      "engine/webrtcmediaengine.h",
+      "engine/webrtcvideocapturer.cc",
+      "engine/webrtcvideocapturer.h",
+      "engine/webrtcvideocapturerfactory.cc",
+      "engine/webrtcvideocapturerfactory.h",
+      "engine/webrtcvideoengine2.cc",
+      "engine/webrtcvideoengine2.h",
+      "engine/webrtcvideoframe.h",
+      "engine/webrtcvoe.h",
+      "engine/webrtcvoiceengine.cc",
+      "engine/webrtcvoiceengine.h",
+    ]
+
+    deps += [
+      "../api:video_frame_api",
+      "../api/audio_codecs:builtin_audio_decoder_factory",
+      "../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/video_capture:video_capture_module",
+      "../modules/video_coding",
+      "../modules/video_coding:webrtc_h264",
+      "../modules/video_coding:webrtc_vp8",
+      "../modules/video_coding:webrtc_vp9",
+      "../video",
+      "../voice_engine",
+    ]
+  }
 }
 
 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 46 matching lines @@
     deps += [
       ":rtc_media",
       ":rtc_media_base",
       "..:webrtc_common",
       "../api:call_api",
       "../api:video_frame_api",
       "../api/video_codecs:video_codecs_api",
       "../base:rtc_base",
       "../base:rtc_base_approved",
       "../base:rtc_base_tests_utils",
-      "../call:call_interfaces",
+      "../call",
       "../test:test_support",
       "//testing/gtest",
     ]
     public_deps += [ "//testing/gmock" ]
   }
 
   config("rtc_media_unittests_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.
@@ skipping 106 matching lines @@
     }
 
     if (is_ios) {
       deps += [ ":rtc_media_unittests_bundle_data" ]
     }
 
     deps += [
       ":rtc_media",
       ":rtc_media_base",
       ":rtc_media_tests_utils",
-      "../api:video_frame_api",
-      "../api/audio_codecs:builtin_audio_decoder_factory",
-      "../api/audio_codecs:builtin_audio_encoder_factory",
       "../api/video_codecs:video_codecs_api",
       "../audio",
       "../base:rtc_base",
       "../base:rtc_base_approved",
       "../base:rtc_base_tests_main",
       "../base:rtc_base_tests_utils",
       "../call:call_interfaces",
-      "../common_video:common_video",
       "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
-      "../modules/audio_processing:audio_processing",
-      "../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",
     ]
+
+    if (rtc_enable_media) {
+      deps += [
+        "../api:video_frame_api",
+        "../api/audio_codecs:builtin_audio_decoder_factory",
+        "../api/audio_codecs:builtin_audio_encoder_factory",
+        "../common_video:common_video",
+        "../modules/audio_processing:audio_processing",
+        "../modules/video_coding:video_coding_utility",
+      ]
+    }
   }
 }