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Add configuration and test for extended RTCP reference time reports to new video api.

Browse Source 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
asapersson@webrtc.org
2014-01-20 08:34:49 +00:00
parent 32c26eb90b
commit efaeda0c76
5 changed files with 112 additions and 8 deletions
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97
webrtc/video/call_tests.cc
View File

@ -127,6 +127,7 @@ class CallTest : public ::testing::Test {
void ReceivesPliAndRecovers(int rtp_history_ms);
void RespectsRtcpMode(newapi::RtcpMode rtcp_mode);
void TestXrReceiverReferenceTimeReport(bool enable_rrtr);
scoped_ptr<Call> sender_call_;
scoped_ptr<Call> receiver_call_;
@ -1051,4 +1052,100 @@ TEST_F(CallTest, ReceiveStreamSendsRemb) {
observer.StopSending();
DestroyStreams();
}
void CallTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
static const int kNumRtcpReportPacketsToObserve = 5;
class RtcpXrObserver : public test::RtpRtcpObserver {
public:
explicit RtcpXrObserver(bool enable_rrtr)
: test::RtpRtcpObserver(kDefaultTimeoutMs),
enable_rrtr_(enable_rrtr),
sent_rtcp_sr_(0),
sent_rtcp_rr_(0),
sent_rtcp_rrtr_(0),
sent_rtcp_dlrr_(0) {}
private:
// Receive stream should send RR packets (and RRTR packets if enabled).
virtual Action OnReceiveRtcp(const uint8_t* packet,
size_t length) OVERRIDE {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::kRtcpNotValidCode) {
if (packet_type == RTCPUtility::kRtcpRrCode) {
++sent_rtcp_rr_;
} else if (
packet_type == RTCPUtility::kRtcpXrReceiverReferenceTimeCode) {
++sent_rtcp_rrtr_;
}
EXPECT_NE(packet_type, RTCPUtility::kRtcpSrCode);
EXPECT_NE(packet_type, RTCPUtility::kRtcpXrDlrrReportBlockItemCode);
packet_type = parser.Iterate();
}
return SEND_PACKET;
}
// Send stream should send SR packets (and DLRR packets if enabled).
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::kRtcpNotValidCode) {
if (packet_type == RTCPUtility::kRtcpSrCode) {
++sent_rtcp_sr_;
} else if (packet_type == RTCPUtility::kRtcpXrDlrrReportBlockItemCode) {
++sent_rtcp_dlrr_;
}
EXPECT_NE(packet_type, RTCPUtility::kRtcpXrReceiverReferenceTimeCode);
packet_type = parser.Iterate();
}
if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve) {
if (enable_rrtr_) {
EXPECT_GT(sent_rtcp_rrtr_, 0);
EXPECT_GT(sent_rtcp_dlrr_, 0);
} else {
EXPECT_EQ(0, sent_rtcp_rrtr_);
EXPECT_EQ(0, sent_rtcp_dlrr_);
}
observation_complete_->Set();
}
return SEND_PACKET;
}
bool enable_rrtr_;
int sent_rtcp_sr_;
int sent_rtcp_rr_;
int sent_rtcp_rrtr_;
int sent_rtcp_dlrr_;
} observer(enable_rrtr);
CreateCalls(Call::Config(observer.SendTransport()),
Call::Config(observer.ReceiveTransport()));
observer.SetReceivers(receiver_call_->Receiver(),
sender_call_->Receiver());
CreateTestConfigs();
receive_config_.rtp.rtcp_mode = newapi::kRtcpReducedSize;
receive_config_.rtp.rtcp_xr.receiver_reference_time_report = enable_rrtr;
CreateStreams();
CreateFrameGenerator();
StartSending();
EXPECT_EQ(kEventSignaled, observer.Wait())
<< "Timed out while waiting for RTCP SR/RR packets to be sent.";
StopSending();
observer.StopSending();
DestroyStreams();
}
TEST_F(CallTest, ReceiverReferenceTimeReportEnabled) {
TestXrReceiverReferenceTimeReport(true);
}
TEST_F(CallTest, ReceiverReferenceTimeReportDisabled) {
TestXrReceiverReferenceTimeReport(false);
}
} // namespace webrtc