Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
| index 0c46e40e1ec38d8414679cde93de00c1ea2c4789..cd0412072573367bbcca5d9c77a08fc77d763ea1 100644 |
| --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
| @@ -25,6 +25,7 @@ |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
| +#include "webrtc/video_send_stream.h" |
|
åsapersson
2017/07/05 15:19:55
remove?
sprang_webrtc
2017/07/05 15:56:47
Done.
|
| namespace webrtc { |
| @@ -335,9 +336,12 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp { |
| const Clock* const clock_; |
| const bool audio_; |
| - int64_t last_process_time_; |
| + |
| + const RtpKeepAliveConfig keepalive_config_; |
| int64_t last_bitrate_process_time_; |
| int64_t last_rtt_process_time_; |
| + int64_t next_process_time_; |
| + int64_t next_keepalive_time_; |
| uint16_t packet_overhead_; |
| // Send side |