Chromium Code Reviews| 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 { |