Index: webrtc/modules/audio_coding/BUILD.gn |
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn |
index 70ecc55dc2aa86d3f2739bdd6d91567fe112be17..11d6297d1dde8bafc4f9210c7ccbb3a780a53113 100644 |
--- a/webrtc/modules/audio_coding/BUILD.gn |
+++ b/webrtc/modules/audio_coding/BUILD.gn |
@@ -913,6 +913,8 @@ 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", |
@@ -924,6 +926,7 @@ rtc_static_library("audio_network_adaptor") { |
"../..:webrtc_common", |
"../../base:rtc_base_approved", |
"../../common_audio", |
+ "../../logging:rtc_event_log_api", |
"../../system_wrappers", |
] |
@@ -934,6 +937,11 @@ 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") { |