commit f6b55092295412a5cf1ffc7e77de0575eaca43b3
author kjellander <kjellander@webrtc.org> Mon Feb 08 07:04:26 2016
committer Commit bot <commit-bot@chromium.org> Mon Feb 08 07:04:33 2016
tree f0cb21bfc6f6955230525bbc927ca68e9a42a733
parent 1afca73055999d1dfcfa61bfc4381eadea26a6d2

Fix GYP and GN references that are invalid in Chromium builds.

There were a couple of GN and GYP references that were incorrect in Chromium builds:
- GN references between WebRTC targets must be using relative paths, not absolute.
- GYP references between WebRTC targets must be using the <(webrtc_root)v variable
  in order to be expanded to the correct path in a Chromium build.

NOTRY=True
TBR=hjon@webrtc.org, hbos@webrtc.org

Review URL: https://codereview.webrtc.org/1681493002

Cr-Commit-Position: refs/heads/master@{#11521}

talk/build/merge_ios_libs.gyp

diff --git a/talk/build/merge_ios_libs.gyp b/talk/build/merge_ios_libs.gyp
index f7e4875..0ec6e85 100644
--- a/talk/build/merge_ios_libs.gyp
+++ b/talk/build/merge_ios_libs.gyp
@@ -37,7 +37,7 @@
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
             '../libjingle.gyp:libjingle_peerconnection_objc',
           ],
-          'sources': ['<(DEPTH)/webrtc/build/no_op.cc',],
+          'sources': ['<(webrtc_root)/build/no_op.cc',],
         },
       ],
     }]

webrtc/api/BUILD.gn

diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 3f47ac8..16c91c7 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -20,8 +20,8 @@
 if (is_ios) {
   source_set("rtc_api_objc") {
     deps = [
-      "//webrtc/base:rtc_base_objc",
-      #"//talk/libjingle:libjingle_peerconnection",
+      "../base:rtc_base_objc",
+      #"../../talk/libjingle:libjingle_peerconnection",
     ]
     cflags = [
       "-fobjc-arc",

webrtc/modules/video_coding/BUILD.gn

diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn
index 4c52a5dbce..489dca4 100644
--- a/webrtc/modules/video_coding/BUILD.gn
+++ b/webrtc/modules/video_coding/BUILD.gn
@@ -150,9 +150,9 @@
       "codecs/h264/h264_encoder_impl.h",
     ]
     deps += [
+      "../../common_video",
       "//third_party/ffmpeg:ffmpeg",
       "//third_party/openh264:encoder",
-      "//webrtc/common_video",
     ]
   }
 }

webrtc/modules/video_coding/codecs/h264/h264.gypi

diff --git a/webrtc/modules/video_coding/codecs/h264/h264.gypi b/webrtc/modules/video_coding/codecs/h264/h264.gypi
index 53f783c..3fe032a 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264.gypi
+++ b/webrtc/modules/video_coding/codecs/h264/h264.gypi
@@ -40,7 +40,7 @@
           'dependencies': [
             '<(DEPTH)/third_party/ffmpeg/ffmpeg.gyp:ffmpeg',
             '<(DEPTH)/third_party/openh264/openh264.gyp:openh264_encoder',
-            '<(DEPTH)/webrtc/common_video/common_video.gyp:common_video',
+            '<(webrtc_root)/common_video/common_video.gyp:common_video',
           ],
           'sources': [
             'h264_decoder_impl.cc',

webrtc/test/test.gyp

diff --git a/webrtc/test/test.gyp b/webrtc/test/test.gyp
index 5b1e24d..492690d 100644
--- a/webrtc/test/test.gyp
+++ b/webrtc/test/test.gyp
@@ -83,8 +83,8 @@
         'rtp_file_writer.h',
       ],
       'dependencies': [
-        '<(DEPTH)/webrtc/common.gyp:webrtc_common',
         '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
       ],
     },