| 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",
|
|
|