Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5791 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_engine/vie_rtp_rtcp_impl.h b/video_engine/vie_rtp_rtcp_impl.h
index 3120bee..1702d63 100644
--- a/video_engine/vie_rtp_rtcp_impl.h
+++ b/video_engine/vie_rtp_rtcp_impl.h
@@ -92,6 +92,8 @@
virtual int SetTransmissionSmoothingStatus(int video_channel, bool enable);
virtual int SetMinTransmitBitrate(int video_channel,
int min_transmit_bitrate_kbps);
+ virtual int SetReservedTransmitBitrate(
+ int video_channel, unsigned int reserved_transmit_bitrate_bps);
virtual int GetReceiveChannelRtcpStatistics(const int video_channel,
RtcpStatistics& basic_stats,
int& rtt_ms) const;