Moved Gn target rtc_event_log to one directory above.
This is done to ensure GN targets are placed in the same directory as of the source files.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2365383004
Cr-Commit-Position: refs/heads/master@{#14411}
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 9ac939d..44488d0 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -313,32 +313,6 @@
}
}
-rtc_static_library("rtc_event_log") {
- sources = [
- "call/rtc_event_log.cc",
- "call/rtc_event_log.h",
- "call/rtc_event_log_helper_thread.cc",
- "call/rtc_event_log_helper_thread.h",
- ]
-
- defines = []
-
- deps = [
- ":webrtc_common",
- "modules/rtp_rtcp",
- ]
-
- if (rtc_enable_protobuf) {
- defines += [ "ENABLE_RTC_EVENT_LOG" ]
- deps += [ ":rtc_event_log_proto" ]
- }
- if (is_clang && !is_nacl) {
- # Suppress warnings from Chrome's Clang plugins.
- # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-}
-
if (rtc_enable_protobuf) {
rtc_static_library("rtc_event_log_parser") {
sources = [
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 32bcba3..648bc85 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -23,7 +23,7 @@
}
deps = [
- "..:rtc_event_log",
+ ":rtc_event_log",
"..:webrtc_common",
"../api:call_api",
"../audio",
@@ -35,6 +35,32 @@
]
}
+rtc_static_library("rtc_event_log") {
+ sources = [
+ "rtc_event_log.cc",
+ "rtc_event_log.h",
+ "rtc_event_log_helper_thread.cc",
+ "rtc_event_log_helper_thread.h",
+ ]
+
+ defines = []
+
+ deps = [
+ "..:webrtc_common",
+ "../modules/rtp_rtcp",
+ ]
+
+ if (rtc_enable_protobuf) {
+ defines += [ "ENABLE_RTC_EVENT_LOG" ]
+ deps += [ "..:rtc_event_log_proto" ]
+ }
+ if (is_clang && !is_nacl) {
+ # Suppress warnings from Chrome's Clang plugins.
+ # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+}
+
if (rtc_include_tests) {
rtc_source_set("call_tests") {
testonly = true
@@ -63,7 +89,7 @@
"rtc_event_log_unittest_helper.cc",
]
deps = [
- "..:rtc_event_log",
+ ":rtc_event_log",
"..:rtc_event_log_parser",
"../modules/rtp_rtcp",
"../system_wrappers:metrics_default",
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 1d16202..bafeffb 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -117,7 +117,7 @@
":audio_network_adaptor",
":neteq",
":rent_a_codec",
- "../..:rtc_event_log",
+ "../../call:rtc_event_log",
]
defines = audio_coding_defines
}
diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn
index 4e5d671..c084e63 100644
--- a/webrtc/test/fuzzers/BUILD.gn
+++ b/webrtc/test/fuzzers/BUILD.gn
@@ -112,7 +112,7 @@
"congestion_controller_feedback_fuzzer.cc",
]
deps = [
- "../../:rtc_event_log",
+ "../../call:rtc_event_log",
"../../modules/congestion_controller/",
]
}
diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn
index 7d9cd40..08206c1 100644
--- a/webrtc/tools/BUILD.gn
+++ b/webrtc/tools/BUILD.gn
@@ -164,8 +164,8 @@
}
defines = [ "ENABLE_RTC_EVENT_LOG" ]
deps = [
- "../:rtc_event_log",
"../:rtc_event_log_parser",
+ "../call:rtc_event_log",
"../modules/congestion_controller:congestion_controller",
"../modules/rtp_rtcp:rtp_rtcp",
"../system_wrappers:system_wrappers_default",
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 14266a6..d411dad 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -55,10 +55,10 @@
}
deps = [
- "..:rtc_event_log",
"..:webrtc_common",
"../base:rtc_base_approved",
"../base:rtc_task_queue",
+ "../call:rtc_event_log",
"../common_video",
"../modules/bitrate_controller",
"../modules/congestion_controller",
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 8ea5328..8450d1d 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -89,10 +89,10 @@
]
deps = [
":level_indicator",
- "..:rtc_event_log",
"..:webrtc_common",
"../api:call_api",
"../base:rtc_base_approved",
+ "../call:rtc_event_log",
"../common_audio",
"../modules/audio_conference_mixer",
"../modules/audio_device",
@@ -244,7 +244,7 @@
":voice_engine",
"//testing/gtest",
"//third_party/gflags",
- "//webrtc:rtc_event_log",
+ "//webrtc/call:rtc_event_log",
"//webrtc/system_wrappers",
"//webrtc/system_wrappers:system_wrappers_default",
"//webrtc/test:test_support",
@@ -271,7 +271,7 @@
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
- "//webrtc/:rtc_event_log",
+ "//webrtc/call:rtc_event_log",
"//webrtc/modules/video_capture",
"//webrtc/system_wrappers",
"//webrtc/system_wrappers/:system_wrappers_default",