Creating webrtc/modules:module_api

webrtc/modules/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("../webrtc.gni")
 import("audio_coding/audio_coding.gni")
@@ skipping 11 matching lines @@
     "media_file",
     "pacing",
     "remote_bitrate_estimator",
     "rtp_rtcp",
     "utility",
     "video_coding",
     "video_processing",
   ]
 }
 
+rtc_source_set("module_api") {
+  sources = [
+    "include/module.h",
+    "include/module_common_types.h",
+  ]
+  deps = [
+    "..:webrtc_common",
+    "../api:video_frame_api",
+    "../base:rtc_base_approved",
+  ]
+}
+
 if (rtc_include_tests) {
   modules_tests_resources = [
     "//resources/audio_coding/testfile32kHz.pcm",
     "//resources/audio_coding/teststereo32kHz.pcm",
     "//resources/foreman_cif.yuv",
     "//resources/paris_qcif.yuv",
   ]
 
   if (is_ios) {
     bundle_data("modules_tests_bundle_data") {
@@ skipping 150 matching lines @@
       testonly = true
       sources = modules_unittests_resources
       outputs = [
         "{{bundle_resources_dir}}/{{source_file_part}}",
       ]
     }
   }
 
   rtc_test("modules_unittests") {
     testonly = true
-
-    deps = []
     defines = []
     sources = [
       "module_common_types_unittest.cc",
     ]
 
     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 += [
+      ":module_api",

[kjellander_webrtc, 2017/04/26 09:36:32]

Ouch, annoying mistake that reminds us to always run additional tests before
submit even if you made the simplest change.

       "../test:test_main",
       "audio_coding:audio_coding_unittests",
       "audio_conference_mixer:audio_conference_mixer_unittests",
       "audio_device:audio_device_unittests",
       "audio_mixer:audio_mixer_unittests",
       "audio_processing:audio_processing_unittests",
       "bitrate_controller:bitrate_controller_unittests",
       "congestion_controller:congestion_controller_unittests",
       "desktop_capture:desktop_capture_unittests",
       "media_file:media_file_unittests",
@@ skipping 16 matching lines @@
       shard_timeout = 900
     }
     if (is_ios) {
       info_plist = "//webrtc/test/ios/Info.plist"
       deps += [ ":modules_unittests_bundle_data" ]
       configs += [ "..:common_objc" ]
       ldflags = [ "-ObjC" ]
     }
   }
 }