| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
|
| index 99465c67f30c3772bd60b03623470d38380880f9..f9d5df68022516d7f4a7b589213d037d80b01c31 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
|
| @@ -13,6 +13,7 @@
|
|
|
| #include <list>
|
| #include <map>
|
| +#include <memory>
|
| #include <utility>
|
| #include <vector>
|
|
|
| @@ -423,8 +424,8 @@ class RTPSender : public RTPSenderInterface {
|
| Bitrate total_bitrate_sent_;
|
|
|
| const bool audio_configured_;
|
| - const rtc::scoped_ptr<RTPSenderAudio> audio_;
|
| - const rtc::scoped_ptr<RTPSenderVideo> video_;
|
| + const std::unique_ptr<RTPSenderAudio> audio_;
|
| + const std::unique_ptr<RTPSenderVideo> video_;
|
|
|
| RtpPacketSender* const paced_sender_;
|
| TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
|
|
|