Index: webrtc/test/call_test.h |
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h |
index 5186afa7533881e625692f20f52962c30403a358..b5271f7d44b6ef9f5bfdbe0acc977857a8fb1d89 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 +163,9 @@ 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: |
+ SingleThreadedTaskQueue task_queue_; // Should be last, to destruct first. |
}; |
class BaseTest : public RtpRtcpObserver { |
@@ -188,8 +192,11 @@ 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( |
+ SingleThreadedTaskQueue* task_queue, |
+ Call* sender_call); |
+ virtual test::PacketTransport* CreateReceiveTransport( |
+ SingleThreadedTaskQueue* task_queue); |
virtual void ModifyVideoConfigs( |
VideoSendStream::Config* send_config, |