Index: webrtc/voice_engine/BUILD.gn |
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn |
index ca774f28006878e345115d37f6171b8c46f6d6e9..be6faabe3957f276565dfbbd4b739cf6f41c8e14 100644 |
--- a/webrtc/voice_engine/BUILD.gn |
+++ b/webrtc/voice_engine/BUILD.gn |
@@ -16,6 +16,7 @@ rtc_static_library("audio_coder") { |
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", |
@@ -39,6 +40,7 @@ rtc_static_library("file_player") { |
"..:webrtc_common", |
"../base:rtc_base_approved", |
"../common_audio", |
+ "../modules:module_api", |
"../modules/media_file", |
] |
@@ -58,6 +60,7 @@ rtc_static_library("file_recorder") { |
"..:webrtc_common", |
"../base:rtc_base_approved", |
"../common_audio", |
+ "../modules:module_api", |
"../modules/media_file:media_file", |
"../system_wrappers", |
] |
@@ -141,6 +144,7 @@ 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/. |
@@ -172,6 +176,7 @@ rtc_static_library("audio_level") { |
"..:webrtc_common", |
"../base:rtc_base_approved", |
"../common_audio", |
+ "../modules:module_api", |
] |
} |
@@ -181,6 +186,7 @@ if (rtc_include_tests) { |
":file_player", |
":voice_engine", |
"../base:rtc_base_approved", |
+ "../modules:module_api", |
"../test:test_common", |
"//testing/gmock", |
"//testing/gtest", |
@@ -244,6 +250,7 @@ if (rtc_include_tests) { |
":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", |