commit 2f1a55583909cdbb234b7da2227afa30150f96d8
author kjellander <kjellander@webrtc.org> Mon Feb 27 15:57:45 2017 -0800
committer Commit bot <commit-bot@chromium.org> Mon Feb 27 23:57:45 2017 +0000
tree f8c4ccbc648ff9d79a1c7c84735a00dfcb678aa9
parent dcb4b437fad571589bdca5daad6e68ee2ee78840

Enable GN check for webrtc/call

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2720503003
Cr-Commit-Position: refs/heads/master@{#16882}

.gn

diff --git a/.gn b/.gn
index db578de..7993d0d 100644
--- a/.gn
+++ b/.gn
@@ -24,6 +24,7 @@
 check_targets = [
   "//webrtc/api/*",
   "//webrtc/audio/*",
+  "//webrtc/call/*",
   "//webrtc/modules/audio_coding/*",
   "//webrtc/modules/audio_conference_mixer/*",
   "//webrtc/modules/audio_device/*",

webrtc/call/BUILD.gn

diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 680ec6b..95db035 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -19,6 +19,15 @@
     "syncable.cc",
     "syncable.h",
   ]
+  deps = [
+    "..:webrtc_common",
+    "../api:audio_mixer_api",
+    "../api:transport_api",
+    "../api/audio_codecs:audio_codecs_api",
+    "../base:rtc_base",
+    "../base:rtc_base_approved",
+    "../modules/audio_coding:audio_encoder_interface",
+  ]
 }
 
 rtc_static_library("call") {
@@ -45,9 +54,13 @@
     "../api:transport_api",
     "../audio",
     "../base:rtc_task_queue",
+    "../logging:rtc_event_log_api",
     "../logging:rtc_event_log_impl",
+    "../modules/bitrate_controller",
     "../modules/congestion_controller",
+    "../modules/pacing",
     "../modules/rtp_rtcp",
+    "../modules/utility",
     "../system_wrappers",
     "../video",
   ]
@@ -65,9 +78,17 @@
     deps = [
       ":call",
       "../base:rtc_base_approved",
+      "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
+      "../modules/bitrate_controller",
+      "../modules/pacing",
+      "../modules/rtp_rtcp",
+      "../system_wrappers",
+      "../test:direct_transport",
       "../test:test_common",
+      "../test:test_support",
+      "../test:video_test_common",
       "//testing/gmock",
       "//testing/gtest",
     ]
@@ -85,6 +106,20 @@
       "rampup_tests.h",
     ]
     deps = [
+      ":call_interfaces",
+      "..:webrtc_common",
+      "../base:rtc_base_approved",
+      "../logging:rtc_event_log_api",
+      "../modules/audio_coding",
+      "../modules/audio_mixer:audio_mixer_impl",
+      "../modules/rtp_rtcp",
+      "../system_wrappers",
+      "../system_wrappers:metrics_default",
+      "../test:direct_transport",
+      "../test:test_support",
+      "../test:video_test_common",
+      "../video",
+      "../voice_engine",
       "//testing/gtest",
       "//webrtc/test:test_common",
     ]