| Index: webrtc/media/BUILD.gn
|
| diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
|
| index 6b18392ee1776400bf897c125d8dc33da2caf170..bb9694f42309d802c3423176dd18bc7facda1486 100644
|
| --- a/webrtc/media/BUILD.gn
|
| +++ b/webrtc/media/BUILD.gn
|
| @@ -231,9 +231,10 @@ rtc_static_library("rtc_media") {
|
| "../call",
|
| "../common_video:common_video",
|
| "../modules/audio_coding:rent_a_codec",
|
| - "../modules/audio_device:audio_device",
|
| - "../modules/audio_mixer:audio_mixer_impl",
|
| - "../modules/audio_processing:audio_processing",
|
| + "../modules/audio_device",
|
| + "../modules/audio_mixer",
|
| + "../modules/audio_processing",
|
| + "../modules/audio_processing/aec_dump",
|
| "../modules/video_capture:video_capture_module",
|
| "../modules/video_coding",
|
| "../modules/video_coding:webrtc_h264",
|
| @@ -244,6 +245,12 @@ rtc_static_library("rtc_media") {
|
| "../video",
|
| "../voice_engine",
|
| ]
|
| +
|
| + if (rtc_enable_protobuf) {
|
| + deps += [ "../modules/audio_processing/aec_dump:aec_dump_impl" ]
|
| + } else {
|
| + deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
|
| + }
|
| }
|
|
|
| if (rtc_include_tests) {
|
|
|