Chromium Code Reviews| Index: webrtc/test/call_test.h |
| diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h |
| index 5186afa7533881e625692f20f52962c30403a358..cf53e32f2012c1a60ace5d92bf6d26bf15da16ba 100644 |
| --- a/webrtc/test/call_test.h |
| +++ b/webrtc/test/call_test.h |
| @@ -23,6 +23,7 @@ |
| #include "webrtc/test/fake_videorenderer.h" |
| #include "webrtc/test/frame_generator_capturer.h" |
| #include "webrtc/test/rtp_rtcp_observer.h" |
| +#include "webrtc/test/single_threaded_task_queue.h" |
| namespace webrtc { |
| @@ -162,6 +163,10 @@ class CallTest : public ::testing::Test { |
| // The audio devices must outlive the voice engines. |
| std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_; |
| std::unique_ptr<test::FakeAudioDevice> fake_recv_audio_device_; |
| + |
| + protected: |
| + // Should be last, to destruct first. |
|
nisse-webrtc
2017/08/21 09:07:07
Explain briefly *why* it has to be destructed firs
eladalon
2017/08/21 10:56:53
Actually, come to think of it, putting this last d
|
| + SingleThreadedTaskQueueForTesting task_queue_; |
| }; |
| class BaseTest : public RtpRtcpObserver { |
| @@ -188,8 +193,11 @@ class BaseTest : public RtpRtcpObserver { |
| RtpTransportControllerSend* controller); |
| virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); |
| - virtual test::PacketTransport* CreateSendTransport(Call* sender_call); |
| - virtual test::PacketTransport* CreateReceiveTransport(); |
| + virtual test::PacketTransport* CreateSendTransport( |
| + SingleThreadedTaskQueueForTesting* task_queue, |
| + Call* sender_call); |
| + virtual test::PacketTransport* CreateReceiveTransport( |
| + SingleThreadedTaskQueueForTesting* task_queue); |
| virtual void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |