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}
This commit is contained in:
Patrik Höglund 2017-11-13 17:04:05 +01:00 committed by Commit Bot
parent 15586d29f6
commit c0e680463a
2 changed files with 14 additions and 5 deletions

View File

@ -86,10 +86,6 @@ if (rtc_include_tests) {
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 @@ if (rtc_include_tests) {
":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",

View File

@ -180,7 +180,6 @@ if (rtc_include_tests) {
"bitrate_allocator_unittest.cc",
"bitrate_estimator_tests.cc",
"call_unittest.cc",
"fake_rtp_transport_controller_send.h",
"flexfec_receive_stream_unittest.cc",
"rtcp_demuxer_unittest.cc",
"rtp_demuxer_unittest.cc",
@ -266,10 +265,14 @@ if (rtc_include_tests) {
testonly = true
sources = [
"fake_rtp_transport_controller_send.h",
"test/mock_rtp_packet_sink_interface.h",
]
deps = [
":rtp_interfaces",
"..:webrtc_common",
"../modules/congestion_controller:congestion_controller",
"../modules/pacing:pacing",
"../test:test_support",
"//testing/gmock",
]