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Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module).

Browse Source 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2383004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
asapersson@webrtc.org
2013-11-20 12:46:11 +00:00
parent 27326b6a42
commit 1ae1d0c471
12 changed files with 115 additions and 59 deletions
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28
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -39,7 +39,7 @@ RtpRtcp::Configuration::Configuration()
rtcp_feedback(NULL),
intra_frame_callback(NULL),
bandwidth_callback(NULL),
rtt_observer(NULL),
rtt_stats(NULL),
audio_messages(NullObjectRtpAudioFeedback()),
remote_bitrate_estimator(NULL),
paced_sender(NULL) {
@ -92,7 +92,9 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
#ifdef MATLAB
, plot1_(NULL),
#endif
rtt_observer_(configuration.rtt_observer) {
rtt_stats_(configuration.rtt_stats),
critical_section_rtt_(CriticalSectionWrapper::CreateCriticalSection()),
rtt_ms_(0) {
send_video_codec_.codecType = kVideoCodecUnknown;
if (default_module_) {
@ -212,8 +214,8 @@ int32_t ModuleRtpRtcpImpl::Process() {
max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
}
// Report the rtt.
if (rtt_observer_ && max_rtt != 0)
rtt_observer_->OnRttUpdate(max_rtt);
if (rtt_stats_ && max_rtt != 0)
rtt_stats_->OnRttUpdate(max_rtt);
}
// Verify receiver reports are delivered and the reported sequence number
@ -240,14 +242,18 @@ int32_t ModuleRtpRtcpImpl::Process() {
// Report rtt from receiver.
if (process_rtt) {
uint16_t rtt_ms;
if (rtt_observer_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
rtt_observer_->OnRttUpdate(rtt_ms);
if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
rtt_stats_->OnRttUpdate(rtt_ms);
}
}
}
// Get processed rtt.
if (process_rtt) {
last_rtt_process_time_ = now;
if (rtt_stats_) {
set_rtt_ms(rtt_stats_->LastProcessedRtt());
}
}
if (rtcp_sender_.TimeToSendRTCPReport()) {
@ -1623,4 +1629,14 @@ void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs);
}
void ModuleRtpRtcpImpl::set_rtt_ms(uint32_t rtt_ms) {
CriticalSectionScoped cs(critical_section_rtt_.get());
rtt_ms_ = rtt_ms;
}
uint32_t ModuleRtpRtcpImpl::rtt_ms() const {
CriticalSectionScoped cs(critical_section_rtt_.get());
return rtt_ms_;
}
} // Namespace webrtc

11
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -19,6 +19,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/gtest_prod_util.h"
#ifdef MATLAB
class MatlabPlot;
@ -383,9 +384,13 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
Clock* clock_;
private:
FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
int64_t RtcpReportInterval();
void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
void set_rtt_ms(uint32_t rtt_ms);
uint32_t rtt_ms() const;
int32_t id_;
const bool audio_;
bool collision_detected_;
@ -414,7 +419,11 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
MatlabPlot* plot1_;
#endif
RtcpRttObserver* rtt_observer_;
RtcpRttStats* rtt_stats_;
// The processed RTT from RtcpRttStats.
scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
uint32_t rtt_ms_;
};
} // namespace webrtc

13
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
View File

@ -18,7 +18,7 @@
namespace webrtc {
namespace {
class RtcpRttStatsTestImpl : public RtcpRttObserver {
class RtcpRttStatsTestImpl : public RtcpRttStats {
public:
RtcpRttStatsTestImpl() : rtt_ms_(0) {}
virtual ~RtcpRttStatsTestImpl() {}
@ -26,6 +26,9 @@ class RtcpRttStatsTestImpl : public RtcpRttObserver {
virtual void OnRttUpdate(uint32_t rtt_ms) {
rtt_ms_ = rtt_ms;
}
virtual uint32_t LastProcessedRtt() const {
return rtt_ms_;
}
uint32_t rtt_ms_;
};
@ -70,7 +73,7 @@ class RtpRtcpImplTest : public ::testing::Test {
configuration.clock = &clock_;
configuration.outgoing_transport = &transport_;
configuration.receive_statistics = receive_statistics_.get();
configuration.rtt_observer = &rtt_stats_;
configuration.rtt_stats = &rtt_stats_;
rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
transport_.SetRtpRtcpModule(rtp_rtcp_impl_.get());
@ -136,10 +139,12 @@ TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
EXPECT_EQ(0, rtp_rtcp_impl_->SendRTCP(kRtcpReport));
// Verify RTT.
EXPECT_EQ(0U, rtt_stats_.rtt_ms_);
EXPECT_EQ(0U, rtt_stats_.LastProcessedRtt());
EXPECT_EQ(0U, rtp_rtcp_impl_->rtt_ms());
rtp_rtcp_impl_->Process();
EXPECT_EQ(100U, rtt_stats_.rtt_ms_);
EXPECT_EQ(100U, rtt_stats_.LastProcessedRtt());
EXPECT_EQ(100U, rtp_rtcp_impl_->rtt_ms());
}
} // namespace webrtc