From c0e680463ab57ded7e19af01193cc128dabc68a3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Patrik=20H=C3=B6glund?= Date: Mon, 13 Nov 2017 17:04:05 +0100 Subject: [PATCH] Fix deps of audio:audio_tests. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:6828 Change-Id: Iae9020fda37fe40221d9a9def38c3afcc387d359 Reviewed-on: https://webrtc-review.googlesource.com/22683 Reviewed-by: Fredrik Solenberg Commit-Queue: Patrik Höglund Cr-Commit-Position: refs/heads/master@{#20666} --- audio/BUILD.gn | 14 ++++++++++---- call/BUILD.gn | 5 ++++- 2 files changed, 14 insertions(+), 5 deletions(-) diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 472211e3f5..038338c03a 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -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", diff --git a/call/BUILD.gn b/call/BUILD.gn index faa611d159..3c834f105b 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -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", ]