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_; |