From 84f8df71afd1d9705fcbae5cf828816876f66156 Mon Sep 17 00:00:00 2001 From: kjellander Date: Wed, 18 May 2016 05:00:50 -0700 Subject: [PATCH] Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ ) Reason for revert: This breaks our Chromium WebRTC FYI bots: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/6173 /mnt/data/b/build/slave/Linux_Builder/build/src/buildtools/linux64/gn gen //out/Release --check -> returned 1 ERROR at //content/renderer/media/rtc_video_decoder.cc:24:11: Can't include this header from here. #include "third_party/webrtc/video_frame.h" ^------------------------------- The target: //content/renderer:renderer is including a file from the target: //third_party/webrtc:webrtc_common The //content/renderer:renderer target should probably be updated to depend on //third_party/webrtc:webrtc_common before relanding this. Original issue's description: > Add missing headers and fix some missing dependencies > > This is the first CL in a series of major cleanup and dependency > corrections needed in order to satisfy 'gn check'. > > BUG=webrtc:4243, webrtc:5589 > NOTRY=True > > Committed: https://crrev.com/7bb6e75723eb64af079446cc6e3ff08c74fe02e4 > Cr-Commit-Position: refs/heads/master@{#12790} TBR=pbos@webrtc.org,henrika@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4243, webrtc:5589 Review-Url: https://codereview.webrtc.org/1989823002 Cr-Commit-Position: refs/heads/master@{#12793} --- webrtc/BUILD.gn | 10 +--------- webrtc/common.gyp | 3 +-- webrtc/webrtc.gyp | 6 +----- 3 files changed, 3 insertions(+), 16 deletions(-) diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index 1b209dda6a..01fa042af3 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -173,14 +173,9 @@ config("common_config") { source_set("webrtc") { sources = [ - "audio_receive_stream.h", - "audio_send_stream.h", - "audio_state.h", "call.h", - "common.h", + "config.h", "transport.h", - "video_receive_stream.h", - "video_send_stream.h", ] defines = [] @@ -241,7 +236,6 @@ source_set("webrtc_common") { "config.h", "engine_configurations.h", "typedefs.h", - "video_frame.h", ] configs += [ ":common_config" ] @@ -277,7 +271,6 @@ source_set("rtc_event_log") { deps = [ ":webrtc_common", - "base:rtc_base_approved", ] if (rtc_enable_protobuf) { @@ -304,7 +297,6 @@ if (rtc_enable_protobuf) { deps = [ ":rtc_event_log_proto", ":webrtc_common", - "base:rtc_base_approved", ] if (is_clang && !is_nacl) { diff --git a/webrtc/common.gyp b/webrtc/common.gyp index 4a111d0587..2970877309 100644 --- a/webrtc/common.gyp +++ b/webrtc/common.gyp @@ -15,11 +15,10 @@ 'audio_sink.h', 'common_types.cc', 'common_types.h', - 'config.cc', 'config.h', + 'config.cc', 'engine_configurations.h', 'typedefs.h', - 'video_frame.h', ], }, ], diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp index 7f9a9ea840..793bf96335 100644 --- a/webrtc/webrtc.gyp +++ b/webrtc/webrtc.gyp @@ -78,7 +78,6 @@ ], 'dependencies': [ 'rtc_event_log_proto', - '<(webrtc_root)/base/base.gyp:rtc_base_approved', ], 'export_dependent_settings': [ 'rtc_event_log_proto', @@ -144,7 +143,7 @@ 'audio_send_stream.h', 'audio_state.h', 'call.h', - 'common.h', + 'config.h', 'transport.h', 'video_receive_stream.h', 'video_send_stream.h', @@ -179,9 +178,6 @@ 'call/rtc_event_log_helper_thread.cc', 'call/rtc_event_log_helper_thread.h', ], - 'dependencies': [ - '<(webrtc_root)/base/base.gyp:rtc_base_approved', - ], 'conditions': [ # If enable_protobuf is defined, we want to compile the protobuf # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.