Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
index 05320f72e1b5a5272bf7c35a2330ce7abe1d6f65..1c7cbd14e793a83d82869e107efc8afe9c420af0 100644 |
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
@@ -387,6 +387,12 @@ class RtpPacketSender { |
int64_t capture_time_ms, |
size_t bytes, |
bool retransmission) = 0; |
+ |
+ // Try to allocate bitrate for sending a retransmission. This may fail if too |
+ // much retransmission is requested in relation to payload. |
+ virtual bool AllocateRetransmissionBitrate(size_t bytes) = 0; |
+ |
+ virtual int CurrentRetransmissionBitrate() = 0; |
danilchap
2016/06/23 12:46:12
maybe call it CurrentRetransmisisonBitrateBps to p
sprang_webrtc
2016/06/28 09:12:33
Done. And removed. :)
|
}; |
class TransportSequenceNumberAllocator { |