Index: webrtc/test/call_test.h |
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h |
index 41a6b5a6c2387e3b07eda1b95140d27bfed77b85..ebc2bb2a544d78a60f2f26de90d7b3de0a1201e2 100644 |
--- a/webrtc/test/call_test.h |
+++ b/webrtc/test/call_test.h |
@@ -82,22 +82,22 @@ class CallTest : public ::testing::Test { |
Clock* const clock_; |
- rtc::scoped_ptr<Call> sender_call_; |
- rtc::scoped_ptr<PacketTransport> send_transport_; |
+ std::unique_ptr<Call> sender_call_; |
+ std::unique_ptr<PacketTransport> send_transport_; |
VideoSendStream::Config video_send_config_; |
VideoEncoderConfig video_encoder_config_; |
VideoSendStream* video_send_stream_; |
AudioSendStream::Config audio_send_config_; |
AudioSendStream* audio_send_stream_; |
- rtc::scoped_ptr<Call> receiver_call_; |
- rtc::scoped_ptr<PacketTransport> receive_transport_; |
+ std::unique_ptr<Call> receiver_call_; |
+ std::unique_ptr<PacketTransport> receive_transport_; |
std::vector<VideoReceiveStream::Config> video_receive_configs_; |
std::vector<VideoReceiveStream*> video_receive_streams_; |
std::vector<AudioReceiveStream::Config> audio_receive_configs_; |
std::vector<AudioReceiveStream*> audio_receive_streams_; |
- rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_; |
+ std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_; |
test::FakeEncoder fake_encoder_; |
std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_; |
size_t num_video_streams_; |
@@ -127,8 +127,8 @@ class CallTest : public ::testing::Test { |
VoiceEngineState voe_recv_; |
// The audio devices must outlive the voice engines. |
- rtc::scoped_ptr<test::FakeAudioDevice> fake_send_audio_device_; |
- rtc::scoped_ptr<test::FakeAudioDevice> fake_recv_audio_device_; |
+ std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_; |
+ std::unique_ptr<test::FakeAudioDevice> fake_recv_audio_device_; |
}; |
class BaseTest : public RtpRtcpObserver { |