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Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )

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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}
This commit is contained in:
kjellander 2016-05-18 05:00:50 -07:00 committed by Commit bot
parent 2f5d600918
commit 84f8df71af
3 changed files with 3 additions and 16 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

10
webrtc/BUILD.gn
View File

@ -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) {

3
webrtc/common.gyp
View File

@ -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',
],
},
],

6
webrtc/webrtc.gyp
View File

@ -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.