Index: webrtc/modules/audio_coding/BUILD.gn |
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn |
index 2cfd159be0af96975b96ec60470624ffd6a86f2f..044d57e6a0713111985b2583cfed74cdfb695011 100644 |
--- a/webrtc/modules/audio_coding/BUILD.gn |
+++ b/webrtc/modules/audio_coding/BUILD.gn |
@@ -914,6 +914,8 @@ rtc_static_library("audio_network_adaptor") { |
"audio_network_adaptor/debug_dump_writer.h", |
"audio_network_adaptor/dtx_controller.cc", |
"audio_network_adaptor/dtx_controller.h", |
+ "audio_network_adaptor/event_log_writer.cc", |
+ "audio_network_adaptor/event_log_writer.h", |
"audio_network_adaptor/fec_controller.cc", |
"audio_network_adaptor/fec_controller.h", |
"audio_network_adaptor/frame_length_controller.cc", |
@@ -925,6 +927,7 @@ rtc_static_library("audio_network_adaptor") { |
"../..:webrtc_common", |
"../../base:rtc_base_approved", |
"../../common_audio", |
+ "../../logging:rtc_event_log_api", |
"../../system_wrappers", |
] |
@@ -935,6 +938,11 @@ rtc_static_library("audio_network_adaptor") { |
] |
defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ] |
} |
+ |
+ if (!build_with_chromium && is_clang) { |
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
+ } |
} |
rtc_static_library("neteq") { |
@@ -1935,6 +1943,7 @@ if (rtc_include_tests) { |
"audio_network_adaptor/channel_controller_unittest.cc", |
"audio_network_adaptor/controller_manager_unittest.cc", |
"audio_network_adaptor/dtx_controller_unittest.cc", |
+ "audio_network_adaptor/event_log_writer_unittest.cc", |
"audio_network_adaptor/fec_controller_unittest.cc", |
"audio_network_adaptor/frame_length_controller_unittest.cc", |
"audio_network_adaptor/mock/mock_controller.h", |