| Index: webrtc/voice_engine/BUILD.gn | 
| diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn | 
| index 429c346f8cece8d745b95518a0e21ede0992b3f9..253eacb59df34b75cd980e800ee8ccb31c416301 100644 | 
| --- a/webrtc/voice_engine/BUILD.gn | 
| +++ b/webrtc/voice_engine/BUILD.gn | 
| @@ -144,13 +144,10 @@ rtc_static_library("voice_engine") { | 
| "../audio/utility:audio_frame_operations", | 
| "../base:rtc_base_approved", | 
| "../base:rtc_task_queue", | 
| -    "../modules:module_api", | 
| - | 
| -    # TODO(nisse): Delete when declaration of RtpTransportController | 
| -    # and related interfaces move to api/. | 
| -    "../call:call_interfaces", | 
| +    "../call:rtp_interfaces", | 
| "../common_audio", | 
| "../logging:rtc_event_log_api", | 
| +    "../modules:module_api", | 
| "../modules/audio_coding:audio_format_conversion", | 
| "../modules/audio_coding:rent_a_codec", | 
| "../modules/audio_conference_mixer", | 
|  |