| Index: webrtc/voice_engine/BUILD.gn
|
| diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
|
| index be6faabe3957f276565dfbbd4b739cf6f41c8e14..ca774f28006878e345115d37f6171b8c46f6d6e9 100644
|
| --- a/webrtc/voice_engine/BUILD.gn
|
| +++ b/webrtc/voice_engine/BUILD.gn
|
| @@ -16,7 +16,6 @@
|
| deps = [
|
| "..:webrtc_common",
|
| "../api/audio_codecs:builtin_audio_decoder_factory",
|
| - "../modules:module_api",
|
| "../modules/audio_coding",
|
| "../modules/audio_coding:audio_encoder_factory_interface",
|
| "../modules/audio_coding:audio_format_conversion",
|
| @@ -40,7 +39,6 @@
|
| "..:webrtc_common",
|
| "../base:rtc_base_approved",
|
| "../common_audio",
|
| - "../modules:module_api",
|
| "../modules/media_file",
|
| ]
|
|
|
| @@ -60,7 +58,6 @@
|
| "..:webrtc_common",
|
| "../base:rtc_base_approved",
|
| "../common_audio",
|
| - "../modules:module_api",
|
| "../modules/media_file:media_file",
|
| "../system_wrappers",
|
| ]
|
| @@ -144,7 +141,6 @@
|
| "../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/.
|
| @@ -176,7 +172,6 @@
|
| "..:webrtc_common",
|
| "../base:rtc_base_approved",
|
| "../common_audio",
|
| - "../modules:module_api",
|
| ]
|
| }
|
|
|
| @@ -186,7 +181,6 @@
|
| ":file_player",
|
| ":voice_engine",
|
| "../base:rtc_base_approved",
|
| - "../modules:module_api",
|
| "../test:test_common",
|
| "//testing/gmock",
|
| "//testing/gtest",
|
| @@ -250,7 +244,6 @@
|
| ":voice_engine",
|
| "..:webrtc_common",
|
| "../base:rtc_base_approved",
|
| - "../modules:module_api",
|
| "../modules/audio_device:audio_device",
|
| "../modules/audio_processing:audio_processing",
|
| "../modules/rtp_rtcp:rtp_rtcp",
|
|
|