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Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.

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Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.

BUG=1298

Review URL: https://webrtc-codereview.appspot.com/1060005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-02-01 14:33:42 +00:00
parent 46d90dcd74
commit b586507986
32 changed files with 523 additions and 393 deletions
Download Patch File Download Diff File Expand all files Collapse all files

75
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@@ -11,7 +11,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include <string.h>
#include <cassert>
#include "webrtc/common_types.h"
@@ -39,8 +38,6 @@ const float kFracMs = 4.294967296E6f;
namespace webrtc {
const WebRtc_UWord16 kDefaultRtt = 200;
static RtpData* NullObjectRtpData() {
static NullRtpData null_rtp_data;
return &null_rtp_data;
@@ -107,6 +104,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
last_packet_timeout_process_time_(
configuration.clock->TimeInMilliseconds()),
last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
packet_overhead_(28), // IPV4 UDP.
critical_section_module_ptrs_(
CriticalSectionWrapper::CreateCriticalSection()),
@@ -258,33 +256,30 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
ProcessDeadOrAliveTimer();
const bool default_instance(child_modules_.empty() ? false : true);
if (!default_instance && rtcp_sender_.TimeToSendRTCPReport()) {
WebRtc_UWord16 max_rtt = 0;
if (!default_instance) {
if (rtcp_sender_.Sending()) {
std::vector<RTCPReportBlock> receive_blocks;
rtcp_receiver_.StatisticsReceived(&receive_blocks);
for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
it != receive_blocks.end(); ++it) {
WebRtc_UWord16 rtt = 0;
rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL);
max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
// Process RTT if we have received a receiver report and we haven't
// processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds.
if (rtcp_receiver_.LastReceivedReceiverReport() >
last_rtt_process_time_ && now >= last_rtt_process_time_ +
kRtpRtcpRttProcessTimeMs) {
last_rtt_process_time_ = now;
std::vector<RTCPReportBlock> receive_blocks;
rtcp_receiver_.StatisticsReceived(&receive_blocks);
uint16_t max_rtt = 0;
for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
it != receive_blocks.end(); ++it) {
uint16_t rtt = 0;
rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL);
max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
}
// Report the rtt.
if (rtt_observer_ && max_rtt != 0)
rtt_observer_->OnRttUpdate(max_rtt);
}
// Report the rtt.
if (rtt_observer_ && max_rtt != 0)
rtt_observer_->OnRttUpdate(max_rtt);
} else {
// No valid RTT estimate, probably since this is a receive only channel.
// Use an estimate set by a send module.
max_rtt = rtcp_receiver_.RTT();
}
if (max_rtt == 0) {
// No own rtt calculation or set rtt, use default value.
max_rtt = kDefaultRtt;
}
// Verify receiver reports are delivered and the reported sequence number is
// increasing.
if (rtcp_sender_.Sending()) {
// Verify receiver reports are delivered and the reported sequence number
// is increasing.
int64_t rtcp_interval = RtcpReportInterval();
if (rtcp_receiver_.RtcpRrTimeout(rtcp_interval)) {
LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
@@ -292,13 +287,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
LOG_F(LS_WARNING) <<
"Timeout: No increase in RTCP RR extended highest sequence number.";
}
}
if (remote_bitrate_) {
// TODO(mflodman) Remove this and let this be propagated by CallStats.
remote_bitrate_->SetRtt(max_rtt);
remote_bitrate_->UpdateEstimate(rtp_receiver_->SSRC(), now);
if (TMMBR()) {
if (remote_bitrate_ && TMMBR()) {
unsigned int target_bitrate = 0;
std::vector<unsigned int> ssrcs;
if (remote_bitrate_->LatestEstimate(&ssrcs, &target_bitrate)) {
@@ -309,7 +299,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
}
}
}
rtcp_sender_.SendRTCP(kRtcpReport);
if (rtcp_sender_.TimeToSendRTCPReport())
rtcp_sender_.SendRTCP(kRtcpReport);
}
if (UpdateRTCPReceiveInformationTimers()) {
@@ -1995,22 +1986,6 @@ void ModuleRtpRtcpImpl::BitrateSent(WebRtc_UWord32* total_rate,
*nack_rate = rtp_sender_.NackOverheadRate();
}
int ModuleRtpRtcpImpl::EstimatedReceiveBandwidth(
WebRtc_UWord32* available_bandwidth) const {
if (remote_bitrate_) {
std::vector<unsigned int> ssrcs;
if (!remote_bitrate_->LatestEstimate(&ssrcs, available_bandwidth)) {
return -1;
}
if (!ssrcs.empty()) {
*available_bandwidth /= ssrcs.size();
}
return 0;
}
// No bandwidth receive-side bandwidth estimation is connected to this module.
return -1;
}
// Bad state of RTP receiver request a keyframe.
void ModuleRtpRtcpImpl::OnRequestIntraFrame() {
RequestKeyFrame();