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Add support for multiple report blocks.

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Use a weighted average of fraction loss for bandwidth estimation.

TEST=trybots and vie_auto_test --automated
BUG=1811
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2198004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4762 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-09-17 07:49:56 +00:00
parent fc10c5c97f
commit 28a331eede
11 changed files with 457 additions and 190 deletions
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27
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@@ -111,10 +111,10 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
configuration.rtcp_feedback);
rtcp_sender_.RegisterSendTransport(configuration.outgoing_transport);
// Make sure that RTCP objects are aware of our SSRC
// Make sure that RTCP objects are aware of our SSRC.
uint32_t SSRC = rtp_sender_.SSRC();
rtcp_sender_.SetSSRC(SSRC);
rtcp_receiver_.SetSSRC(SSRC);
SetRtcpReceiverSsrcs(SSRC);
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s created", __FUNCTION__);
}
@@ -261,6 +261,8 @@ int32_t ModuleRtpRtcpImpl::Process() {
int32_t ModuleRtpRtcpImpl::SetRTXSendStatus(RtxMode mode, bool set_ssrc,
uint32_t ssrc) {
rtp_sender_.SetRTXStatus(mode, set_ssrc, ssrc);
return 0;
}
@@ -410,8 +412,9 @@ int32_t ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetSSRC(%d)", ssrc);
rtp_sender_.SetSSRC(ssrc);
rtcp_receiver_.SetSSRC(ssrc);
rtcp_sender_.SetSSRC(ssrc);
SetRtcpReceiverSsrcs(ssrc);
return 0; // TODO(pwestin): change to void.
}
@@ -510,8 +513,9 @@ int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
// Make sure that RTCP objects are aware of our SSRC (it could have changed
// Due to collision)
uint32_t SSRC = rtp_sender_.SSRC();
rtcp_receiver_.SetSSRC(SSRC);
rtcp_sender_.SetSSRC(SSRC);
SetRtcpReceiverSsrcs(SSRC);
return 0;
}
return 0;
@@ -1459,7 +1463,7 @@ void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
}
// Change local SSRC and inform all objects about the new SSRC.
rtcp_sender_.SetSSRC(new_ssrc);
rtcp_receiver_.SetSSRC(new_ssrc);
SetRtcpReceiverSsrcs(new_ssrc);
}
}
@@ -1587,4 +1591,17 @@ int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
else
return RTCP_INTERVAL_VIDEO_MS;
}
void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
std::set<uint32_t> ssrcs;
ssrcs.insert(main_ssrc);
RtxMode rtx_mode = kRtxOff;
uint32_t rtx_ssrc = 0;
int rtx_payload_type = 0;
rtp_sender_.RTXStatus(&rtx_mode, &rtx_ssrc, &rtx_payload_type);
if (rtx_mode != kRtxOff)
ssrcs.insert(rtx_ssrc);
rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs);
}
} // Namespace webrtc