| Index: webrtc/audio/BUILD.gn
|
| diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
|
| index 2b7d06fcfa988fdea9648938391d138cad8654de..1577316cd4b59d6fcd20697d9f3851294b7febe7 100644
|
| --- a/webrtc/audio/BUILD.gn
|
| +++ b/webrtc/audio/BUILD.gn
|
| @@ -37,6 +37,8 @@ rtc_static_library("audio") {
|
| "../api:call_api",
|
| "../api/audio_codecs:audio_codecs_api",
|
| "../api/audio_codecs:builtin_audio_encoder_factory",
|
| + "../base:rtc_base_approved",
|
| + "../base:rtc_task_queue",
|
| "../call:call_interfaces",
|
| "../call:rtp_interfaces",
|
| "../common_audio",
|
| @@ -48,8 +50,6 @@ rtc_static_library("audio") {
|
| "../modules/pacing:pacing",
|
| "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
|
| "../modules/rtp_rtcp:rtp_rtcp",
|
| - "../rtc_base:rtc_base_approved",
|
| - "../rtc_base:rtc_task_queue",
|
| "../system_wrappers",
|
| "../voice_engine",
|
| ]
|
| @@ -77,14 +77,14 @@ if (rtc_include_tests) {
|
| deps = [
|
| ":audio",
|
| "../api:mock_audio_mixer",
|
| + "../base:rtc_base_approved",
|
| + "../base:rtc_task_queue",
|
| "../call:rtp_receiver",
|
| "../modules/audio_device:mock_audio_device",
|
| "../modules/audio_mixer:audio_mixer_impl",
|
| "../modules/congestion_controller:congestion_controller",
|
| "../modules/congestion_controller:mock_congestion_controller",
|
| "../modules/pacing:pacing",
|
| - "../rtc_base:rtc_base_approved",
|
| - "../rtc_base:rtc_task_queue",
|
| "../test:test_common",
|
| "../test:test_support",
|
| "utility:utility_tests",
|
|
|