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Unified Diff: trunk/src/third_party/libjingle/libjingle.gyp

Issue 13009007: Revert 190016 "Reland libjingle.gyp refactoring change." (Closed) Base URL: svn://svn.chromium.org/chrome/
Patch Set: Created 7 years, 9 months ago
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Index: trunk/src/third_party/libjingle/libjingle.gyp
===================================================================
--- trunk/src/third_party/libjingle/libjingle.gyp (revision 190020)
+++ trunk/src/third_party/libjingle/libjingle.gyp (working copy)
@@ -11,7 +11,7 @@
'libjingle_additional_deps%': [],
'libjingle_peerconnection_additional_deps%': [],
'libjingle_source%': "source",
- 'libpeer_target_type%': '<(component)',
+ 'libpeer_target_type%': 'static_library',
},
'target_defaults': {
'defines': [
@@ -490,9 +490,8 @@
],
'dependencies': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+ 'libjingle_p2p_constants',
'<@(libjingle_additional_deps)',
- 'libjingle_media_base_constants',
- 'libjingle_p2p_base_constants',
],
'export_dependent_settings': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
@@ -567,12 +566,58 @@
'<(libjingle_source)/talk/base/opensslstreamadapter.cc',
],
}],
- ['enable_webrtc==1', {
- 'dependencies': [
- '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
- '<(DEPTH)/third_party/webrtc/modules/modules.gyp:media_file',
- '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture_module',
- ],
+ ],
+ }, # target libjingle
+ # This has to be is a separate project due to a bug in MSVS 2008 and the
+ # current toolset on android. The problem is that we have two files named
+ # "constants.cc" and MSVS/android doesn't handle this properly.
+ # GYP currently has guards to catch this, so if you want to remove it,
+ # run GYP and if GYP has removed the validation check, then we can assume
+ # that the toolchains have been fixed (we currently use VS2010 and later,
+ # so VS2008 isn't a concern anymore).
+ {
+ 'target_name': 'libjingle_p2p_constants',
+ 'type': 'static_library',
+ 'sources': [
+ '<(libjingle_source)/talk/p2p/base/constants.cc',
+ '<(libjingle_source)/talk/p2p/base/constants.h',
+ ],
+ }, # target libjingle_p2p_constants
+ {
+ 'target_name': 'peerconnection_server',
+ 'type': 'executable',
+ 'sources': [
+ '<(libjingle_source)/talk/examples/peerconnection/server/data_socket.cc',
+ '<(libjingle_source)/talk/examples/peerconnection/server/data_socket.h',
+ '<(libjingle_source)/talk/examples/peerconnection/server/main.cc',
+ '<(libjingle_source)/talk/examples/peerconnection/server/peer_channel.cc',
+ '<(libjingle_source)/talk/examples/peerconnection/server/peer_channel.h',
+ '<(libjingle_source)/talk/examples/peerconnection/server/utils.cc',
+ '<(libjingle_source)/talk/examples/peerconnection/server/utils.h',
+ ],
+ 'include_dirs': [
+ '<(libjingle_source)',
+ ],
+ 'dependencies': [
+ 'libjingle',
+ ],
+ # TODO(jschuh): crbug.com/167187 fix size_t to int truncations.
+ 'msvs_disabled_warnings': [ 4309, ],
+ }, # target peerconnection_server
+ ],
+ 'conditions': [
+ ['enable_webrtc==1', {
+ 'targets': [
+ {
+ 'target_name': 'libpeerconnection',
+ 'type': '<(libpeer_target_type)',
+ 'all_dependent_settings': {
+ 'conditions': [
+ ['"<(libpeer_target_type)"=="static_library"', {
+ 'defines': [ 'LIBPEERCONNECTION_LIB=1' ],
+ }],
+ ],
+ },
'sources': [
'<(libjingle_source)/talk/app/webrtc/audiotrack.cc',
'<(libjingle_source)/talk/app/webrtc/audiotrack.h',
@@ -628,6 +673,8 @@
'<(libjingle_source)/talk/media/base/capturerenderadapter.h',
'<(libjingle_source)/talk/media/base/codec.cc',
'<(libjingle_source)/talk/media/base/codec.h',
+ '<(libjingle_source)/talk/media/base/constants.cc',
+ '<(libjingle_source)/talk/media/base/constants.h',
'<(libjingle_source)/talk/media/base/cryptoparams.h',
'<(libjingle_source)/talk/media/base/filemediaengine.cc',
'<(libjingle_source)/talk/media/base/filemediaengine.h',
@@ -656,10 +703,14 @@
'<(libjingle_source)/talk/media/webrtc/webrtcpassthroughrender.cc',
'<(libjingle_source)/talk/media/webrtc/webrtcvideocapturer.cc',
'<(libjingle_source)/talk/media/webrtc/webrtcvideocapturer.h',
+ '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine.cc',
+ '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine.h',
'<(libjingle_source)/talk/media/webrtc/webrtcvideoframe.cc',
'<(libjingle_source)/talk/media/webrtc/webrtcvideoframe.h',
'<(libjingle_source)/talk/media/webrtc/webrtcvie.h',
'<(libjingle_source)/talk/media/webrtc/webrtcvoe.h',
+ '<(libjingle_source)/talk/media/webrtc/webrtcvoiceengine.cc',
+ '<(libjingle_source)/talk/media/webrtc/webrtcvoiceengine.h',
'<(libjingle_source)/talk/session/media/audiomonitor.cc',
'<(libjingle_source)/talk/session/media/audiomonitor.h',
'<(libjingle_source)/talk/session/media/call.cc',
@@ -696,6 +747,19 @@
'<(libjingle_source)/talk/session/tunnel/tunnelsessionclient.h',
],
'conditions': [
+ ['"<(libpeer_target_type)"=="shared_library"', {
+ # Used to control symbol export/import.
+ 'defines': [ 'LIBPEERCONNECTION_IMPLEMENTATION=1' ],
+ }],
+ ['OS=="win" and "<(libpeer_target_type)"=="shared_library"', {
+ 'link_settings': {
+ 'libraries': [
+ '-lsecur32.lib',
+ '-lcrypt32.lib',
+ '-liphlpapi.lib',
+ ],
+ },
+ }],
['enabled_libjingle_device_manager==1', {
'sources!': [
'<(libjingle_source)/talk/media/devices/dummydevicemanager.cc',
@@ -770,89 +834,9 @@
],
}],
],
- }],
- ],
- }, # target libjingle
- # This has to be is a separate project due to a bug in MSVS 2008 and the
- # current toolset on android. The problem is that we have two files named
- # "constants.cc" and MSVS/android doesn't handle this properly.
- # GYP currently has guards to catch this, so if you want to remove it,
- # run GYP and if GYP has removed the validation check, then we can assume
- # that the toolchains have been fixed (we currently use VS2010 and later,
- # so VS2008 isn't a concern anymore).
- {
- 'target_name': 'libjingle_p2p_base_constants',
- 'type': 'static_library',
- 'sources': [
- '<(libjingle_source)/talk/p2p/base/constants.cc',
- '<(libjingle_source)/talk/p2p/base/constants.h',
- ],
- }, # target libjingle_p2p_base_constants
- {
- 'target_name': 'libjingle_media_base_constants',
- 'type': 'static_library',
- 'sources': [
- '<(libjingle_source)/talk/media/base/constants.cc',
- '<(libjingle_source)/talk/media/base/constants.h',
- ],
- }, # target libjingle_media_base_constants
- {
- 'target_name': 'peerconnection_server',
- 'type': 'executable',
- 'sources': [
- '<(libjingle_source)/talk/examples/peerconnection/server/data_socket.cc',
- '<(libjingle_source)/talk/examples/peerconnection/server/data_socket.h',
- '<(libjingle_source)/talk/examples/peerconnection/server/main.cc',
- '<(libjingle_source)/talk/examples/peerconnection/server/peer_channel.cc',
- '<(libjingle_source)/talk/examples/peerconnection/server/peer_channel.h',
- '<(libjingle_source)/talk/examples/peerconnection/server/utils.cc',
- '<(libjingle_source)/talk/examples/peerconnection/server/utils.h',
- ],
- 'include_dirs': [
- '<(libjingle_source)',
- ],
- 'dependencies': [
- 'libjingle',
- ],
- # TODO(jschuh): crbug.com/167187 fix size_t to int truncations.
- 'msvs_disabled_warnings': [ 4309, ],
- }, # target peerconnection_server
- ],
- 'conditions': [
- ['enable_webrtc==1', {
- 'targets': [
- {
- 'target_name': 'libpeerconnection',
- 'type': '<(libpeer_target_type)',
- 'all_dependent_settings': {
- 'conditions': [
- ['"<(libpeer_target_type)"=="static_library"', {
- 'defines': [ 'LIBPEERCONNECTION_LIB=1' ],
- }],
- ],
- },
- 'sources': [
- '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine.cc',
- '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine.h',
- '<(libjingle_source)/talk/media/webrtc/webrtcvoiceengine.cc',
- '<(libjingle_source)/talk/media/webrtc/webrtcvoiceengine.h',
- ],
- 'conditions': [
- ['"<(libpeer_target_type)"=="shared_library"', {
- # Used to control symbol export/import.
- 'defines': [ 'LIBPEERCONNECTION_IMPLEMENTATION=1' ],
- }],
- ['OS=="win" and "<(libpeer_target_type)"=="shared_library"', {
- 'link_settings': {
- 'libraries': [
- '-lsecur32.lib',
- '-lcrypt32.lib',
- '-liphlpapi.lib',
- ],
- },
- }],
- ],
'dependencies': [
+ '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
+ '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture_module',
'<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_render_module',
'<(DEPTH)/third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(DEPTH)/third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
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