Fix deps of audio:audio_tests.
Bug: webrtc:6828
Change-Id: Iae9020fda37fe40221d9a9def38c3afcc387d359
Reviewed-on: https://webrtc-review.googlesource.com/22683
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20666}diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 472211e..038338c 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -86,10 +86,6 @@
rtc_source_set("audio_tests") {
testonly = true
- # TODO(kjellander): Remove (bugs.webrtc.org/6828)
- # This needs remote_bitrate_estimator to be moved to webrtc/api first.
- check_includes = false
-
sources = [
"audio_receive_stream_unittest.cc",
"audio_send_stream_tests.cc",
@@ -101,16 +97,26 @@
":audio",
":audio_end_to_end_test",
"../api:mock_audio_mixer",
+ "../call:mock_rtp_interfaces",
+ "../call:rtp_interfaces",
"../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:mock_paced_sender",
"../modules/pacing:pacing",
+ "../modules/rtp_rtcp:mock_rtp_rtcp",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:rtc_base_approved",
+ "../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_task_queue",
+ "../system_wrappers:system_wrappers",
+ "../test:audio_codec_mocks",
+ "../test:rtp_test_utils",
"../test:test_common",
"../test:test_support",
+ "../voice_engine",
"utility:utility_tests",
"//testing/gmock",
"//testing/gtest",