Chromium Code Reviews| Index: call/BUILD.gn |
| diff --git a/call/BUILD.gn b/call/BUILD.gn |
| index 706d502b213577e0eb16732ad0ef9647c73fbdad..e008ef278fdec214d241b96656ca2dfda234c9a3 100644 |
| --- a/call/BUILD.gn |
| +++ b/call/BUILD.gn |
| @@ -88,6 +88,15 @@ rtc_source_set("rtp_sender") { |
| ] |
| } |
| +rtc_source_set("bitrate_allocator_header") { |
|
mbonadei
2017/09/19 09:00:42
Is it possible to move this into the "call" target
kjellander_webrtc
2017/09/19 16:01:43
Yeah, let's ask owners for advice here.
|
| + sources = [ |
| + "bitrate_allocator.h", |
| + ] |
| + deps = [ |
| + "../rtc_base:sequenced_task_checker", |
| + ] |
| +} |
| + |
| rtc_static_library("call") { |
| sources = [ |
| "bitrate_allocator.cc", |
| @@ -110,6 +119,7 @@ rtc_static_library("call") { |
| ] |
| deps = [ |
| + ":bitrate_allocator_header", |
| ":call_interfaces", |
| ":rtp_interfaces", |
| ":rtp_receiver", |
| @@ -168,6 +178,7 @@ 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", |
| @@ -175,6 +186,7 @@ if (rtc_include_tests) { |
| "rtx_receive_stream_unittest.cc", |
| ] |
| deps = [ |
| + ":bitrate_allocator_header", |
| ":call", |
| ":mock_rtp_interfaces", |
| ":rtp_interfaces", |
| @@ -187,6 +199,7 @@ if (rtc_include_tests) { |
| "../modules/audio_device:mock_audio_device", |
| "../modules/audio_mixer", |
| "../modules/bitrate_controller", |
| + "../modules/congestion_controller:congestion_controller", |
| "../modules/congestion_controller:mock_congestion_controller", |
| "../modules/pacing", |
| "../modules/pacing:mock_paced_sender", |