Index: webrtc/modules/audio_coding/BUILD.gn |
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn |
index efbf7e051b60c8be826afa53a4fe2980e2e4749a..c7e091a35ba0d36c204e512353649c0d62908354 100644 |
--- a/webrtc/modules/audio_coding/BUILD.gn |
+++ b/webrtc/modules/audio_coding/BUILD.gn |
@@ -849,6 +849,8 @@ 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", |
@@ -858,6 +860,7 @@ rtc_static_library("audio_network_adaptor") { |
deps = [ |
"../..:webrtc_common", |
+ "../../logging:rtc_event_log_api", |
"../../system_wrappers", |
] |
@@ -868,6 +871,11 @@ 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" ] |
+ } |
} |
config("neteq_config") { |