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Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss.

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R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5315 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2013-12-18 09:44:53 +00:00
parent 8ae72560dd
commit 1fa41be66a
1 changed files with 28 additions and 56 deletions
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84
webrtc/video/call_tests.cc
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@ -143,16 +143,16 @@ class CallTest : public ::testing::Test {
};
class NackObserver : public test::RtpRtcpObserver {
static const int kNumberOfNacksToObserve = 4;
static const int kInverseProbabilityToStartLossBurst = 20;
static const int kMaxLossBurst = 10;
static const int kNumberOfNacksToObserve = 2;
static const int kLossBurstSize = 2;
static const int kPacketsBetweenLossBursts = 9;
public:
NackObserver()
: test::RtpRtcpObserver(kLongTimeoutMs),
rtp_parser_(RtpHeaderParser::Create()),
drop_burst_count_(0),
sent_rtp_packets_(0),
packets_left_to_drop_(0),
nacks_left_(kNumberOfNacksToObserve) {}
private:
@ -166,31 +166,29 @@ class NackObserver : public test::RtpRtcpObserver {
if (dropped_packets_.find(header.sequenceNumber) !=
dropped_packets_.end()) {
retransmitted_packets_.insert(header.sequenceNumber);
if (nacks_left_ == 0 &&
retransmitted_packets_.size() == dropped_packets_.size()) {
observation_complete_->Set();
}
return SEND_PACKET;
}
// Enough NACKs received, stop dropping packets.
if (nacks_left_ == 0) {
++sent_rtp_packets_;
return SEND_PACKET;
}
// Still dropping packets.
if (drop_burst_count_ > 0) {
--drop_burst_count_;
dropped_packets_.insert(header.sequenceNumber);
return DROP_PACKET;
}
// Should we start dropping packets?
if (sent_rtp_packets_ > 0 &&
rand() % kInverseProbabilityToStartLossBurst == 0) {
drop_burst_count_ = rand() % kMaxLossBurst;
dropped_packets_.insert(header.sequenceNumber);
return DROP_PACKET;
}
++sent_rtp_packets_;
// Enough NACKs received, stop dropping packets.
if (nacks_left_ == 0)
return SEND_PACKET;
// Check if it's time for a new loss burst.
if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
packets_left_to_drop_ = kLossBurstSize;
if (packets_left_to_drop_ > 0) {
--packets_left_to_drop_;
dropped_packets_.insert(header.sequenceNumber);
return DROP_PACKET;
}
return SEND_PACKET;
}
@ -198,50 +196,24 @@ class NackObserver : public test::RtpRtcpObserver {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
bool received_nack = false;
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::kRtcpNotValidCode) {
if (packet_type == RTCPUtility::kRtcpRtpfbNackCode)
received_nack = true;
if (packet_type == RTCPUtility::kRtcpRtpfbNackCode) {
--nacks_left_;
break;
}
packet_type = parser.Iterate();
}
if (received_nack) {
ReceivedNack();
} else {
RtcpWithoutNack();
}
return SEND_PACKET;
}
private:
void ReceivedNack() {
if (nacks_left_ > 0)
--nacks_left_;
rtcp_without_nack_count_ = 0;
}
void RtcpWithoutNack() {
if (nacks_left_ > 0)
return;
++rtcp_without_nack_count_;
// All packets retransmitted and no recent NACKs.
if (dropped_packets_.size() == retransmitted_packets_.size() &&
rtcp_without_nack_count_ >= kRequiredRtcpsWithoutNack) {
observation_complete_->Set();
}
}
scoped_ptr<RtpHeaderParser> rtp_parser_;
std::set<uint16_t> dropped_packets_;
std::set<uint16_t> retransmitted_packets_;
int drop_burst_count_;
uint64_t sent_rtp_packets_;
int packets_left_to_drop_;
int nacks_left_;
int rtcp_without_nack_count_;
static const int kRequiredRtcpsWithoutNack = 2;
};
TEST_F(CallTest, UsesTraceCallback) {