Make some dependencies more flexible.
BUG=none
TEST=trybot
Review URL: https://webrtc-codereview.appspot.com/728005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2583 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/build/common.gypi b/src/build/common.gypi
index 2bb8775..284a5a8 100644
--- a/src/build/common.gypi
+++ b/src/build/common.gypi
@@ -9,26 +9,42 @@
# This file contains common settings for building WebRTC components.
{
+ # Nesting is required in order to use variables for setting other variables.
'variables': {
- # These variables need to be nested in order to use them in a conditions
- # block to set other variables.
'variables': {
'variables': {
- # This will be set to zero in the supplement.gypi triggered by a gclient
- # hook in the standalone build.
- 'build_with_chromium%': 1,
+ 'variables': {
+ # This will be set to zero in the supplement.gypi triggered by a
+ # gclient hook in the standalone build.
+ 'build_with_chromium%': 1,
+ },
+ 'build_with_chromium%': '<(build_with_chromium)',
+
+ 'conditions': [
+ ['build_with_chromium==1', {
+ 'webrtc_root%': '<(DEPTH)/third_party/webrtc',
+ }, {
+ 'webrtc_root%': '<(DEPTH)/src',
+ }],
+ ],
},
'build_with_chromium%': '<(build_with_chromium)',
+ 'webrtc_root%': '<(webrtc_root)',
+
'conditions': [
['build_with_chromium==1', {
- 'webrtc_root%': '<(DEPTH)/third_party/webrtc',
+ 'include_tests%': 0,
}, {
- 'webrtc_root%': '<(DEPTH)/src',
+ 'include_tests%': 1,
}],
],
+
+ 'webrtc_vp8_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp8',
},
'build_with_chromium%': '<(build_with_chromium)',
+ 'include_tests%': '<(include_tests)',
'webrtc_root%': '<(webrtc_root)',
+ 'webrtc_vp8_dir%': '<(webrtc_vp8_dir)',
# The Chromium common.gypi we use treats all gyp files without
# chromium_code==1 as third party code. This disables many of the
@@ -54,7 +70,6 @@
'build_libjpeg%': 1,
'build_libyuv%': 1,
- 'webrtc_vp8_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp8',
'libyuv_dir%': '<(DEPTH)/third_party/libyuv',
'conditions': [
@@ -80,10 +95,7 @@
# Disable the use of protocol buffers in production code.
'enable_protobuf%': 0,
-
- # Don't include tests for Chromium builds.
- 'include_tests%': 0,
- }, { # Settings for the standalone (not-in-Chromium) build.
+ }, { # Settings for the standalone (not-in-Chromium) build.
'include_pulse_audio%': 1,
'include_internal_audio_device%': 1,
@@ -96,8 +108,6 @@
'enable_protobuf%': 1,
- 'include_tests%': 1,
-
# TODO(andrew): For now, disable the Chrome plugins, which causes a
# flood of chromium-style warnings. Investigate enabling them:
# http://code.google.com/p/webrtc/issues/detail?id=163
diff --git a/src/voice_engine/voice_engine_core.gypi b/src/voice_engine/voice_engine_core.gypi
index 10a8d05..4825072 100644
--- a/src/voice_engine/voice_engine_core.gypi
+++ b/src/voice_engine/voice_engine_core.gypi
@@ -7,22 +7,27 @@
# be found in the AUTHORS file in the root of the source tree.
{
+ 'variables': {
+ 'voice_engine_dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
+ '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
+ '<(webrtc_root)/modules/modules.gyp:audio_device',
+ '<(webrtc_root)/modules/modules.gyp:audio_processing',
+ '<(webrtc_root)/modules/modules.gyp:media_file',
+ '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
+ '<(webrtc_root)/modules/modules.gyp:udp_transport',
+ '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ },
'targets': [
{
'target_name': 'voice_engine_core',
'type': '<(library)',
'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
- '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
- '<(webrtc_root)/modules/modules.gyp:audio_device',
- '<(webrtc_root)/modules/modules.gyp:audio_processing',
- '<(webrtc_root)/modules/modules.gyp:media_file',
- '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
- '<(webrtc_root)/modules/modules.gyp:udp_transport',
- '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<@(voice_engine_dependencies)',
],
'include_dirs': [
'include',
diff --git a/webrtc.gyp b/webrtc.gyp
index c729eb8..7d7c051 100644
--- a/webrtc.gyp
+++ b/webrtc.gyp
@@ -8,27 +8,32 @@
{
'includes': [ 'src/build/common.gypi', ],
+ 'variables': {
+ 'webrtc_all_dependencies': [
+ 'src/common_audio/common_audio.gyp:*',
+ 'src/common_video/common_video.gyp:*',
+ 'src/modules/modules.gyp:*',
+ 'src/system_wrappers/source/system_wrappers.gyp:*',
+ 'src/video_engine/video_engine.gyp:*',
+ 'src/voice_engine/voice_engine.gyp:*',
+ '<(webrtc_vp8_dir)/vp8.gyp:*',
+ ],
+ 'conditions': [
+ ['include_tests==1', {
+ 'webrtc_all_dependencies': [
+ 'src/test/metrics.gyp:*',
+ 'src/test/test.gyp:*',
+ 'tools/e2e_quality/e2e_quality.gyp:*',
+ ],
+ }],
+ ],
+ },
'targets': [
{
'target_name': 'All',
'type': 'none',
'dependencies': [
- 'src/common_audio/common_audio.gyp:*',
- 'src/common_video/common_video.gyp:*',
- 'src/modules/modules.gyp:*',
- 'src/system_wrappers/source/system_wrappers.gyp:*',
- 'src/video_engine/video_engine.gyp:*',
- 'src/voice_engine/voice_engine.gyp:*',
- '<(webrtc_vp8_dir)/vp8.gyp:*'
- ],
- 'conditions': [
- ['include_tests==1', {
- 'dependencies': [
- 'src/test/metrics.gyp:*',
- 'src/test/test.gyp:*',
- 'tools/e2e_quality/e2e_quality.gyp:*',
- ],
- }],
+ '<@(webrtc_all_dependencies)',
],
},
],
