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Adds integration test for RTX and fixes bugs found.

Browse Source 
BUG=1811
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2013-05-23 13:48:22 +00:00
parent 5c58f63d3f
commit c74c3c2447
8 changed files with 92 additions and 19 deletions
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1
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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@ -510,7 +510,6 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) {
TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::ReSendPacket",
"timestamp", rtp_header.header.timestamp,
"seqnum", rtp_header.header.sequenceNumber);
if (paced_sender_) {
if (!paced_sender_->SendPacket(PacedSender::kHighPriority,
rtp_header.header.ssrc,

1
webrtc/modules/video_coding/main/source/jitter_buffer.cc
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@ -627,7 +627,6 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
int64_t now_ms = clock_->TimeInMilliseconds();
VCMFrameBufferEnum buffer_return = kSizeError;
VCMFrameBufferEnum ret = kSizeError;
VCMFrameBuffer* frame = NULL;
const VCMFrameBufferEnum error = GetFrame(packet, &frame);
if (error != kNoError) {

3
webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
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@ -255,8 +255,9 @@ void TestFullStack(const TbInterfaces& interfaces,
int32_t num_rtp_packets = 0;
int32_t num_dropped_packets = 0;
int32_t num_rtcp_packets = 0;
std::map<uint8_t, int> packet_counters;
external_transport.GetStats(num_rtp_packets, num_dropped_packets,
num_rtcp_packets);
num_rtcp_packets, &packet_counters);
ViETest::Log("RTP packets : %5d", num_rtp_packets);
ViETest::Log("Dropped packets: %5d", num_dropped_packets);
ViETest::Log("RTCP packets : %5d", num_rtcp_packets);

80
webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
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@ -160,17 +160,83 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
EXPECT_STRCASEEQ(sendCName, remoteCName);
}
//
// RTX
//
unsigned short recFractionsLost = 0;
unsigned int recCumulativeLost = 0;
unsigned int recExtendedMax = 0;
unsigned int recJitter = 0;
int recRttMs = 0;
unsigned int sentTotalBitrate = 0;
unsigned int sentVideoBitrate = 0;
unsigned int sentFecBitrate = 0;
unsigned int sentNackBitrate = 0;
ViETest::Log("Testing NACK over RTX\n");
EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
myTransport.ClearStats();
const uint8_t kRtxPayloadType = 96;
EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
EXPECT_EQ(0, ViE.rtp_rtcp->SetRtxSendPayloadType(tbChannel.videoChannel,
kRtxPayloadType));
EXPECT_EQ(0, ViE.rtp_rtcp->SetRtxReceivePayloadType(tbChannel.videoChannel,
kRtxPayloadType));
EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(tbChannel.videoChannel, 1234,
webrtc::kViEStreamTypeRtx, 0));
EXPECT_EQ(0, ViE.rtp_rtcp->SetRemoteSSRCType(tbChannel.videoChannel,
webrtc::kViEStreamTypeRtx,
1234));
EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
tbChannel.videoChannel, startSequenceNumber));
EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
// Make sure the first key frame gets through.
AutoTestSleep(100);
const int kPacketLossRate = 20;
NetworkParameters network;
network.packet_loss_rate = kPacketLossRate;
network.loss_model = kUniformLoss;
myTransport.SetNetworkParameters(network);
AutoTestSleep(kAutoTestSleepTimeMs);
EXPECT_EQ(0, ViE.rtp_rtcp->GetReceivedRTCPStatistics(
tbChannel.videoChannel, recFractionsLost, recCumulativeLost,
recExtendedMax, recJitter, recRttMs));
EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
sentFecBitrate, sentNackBitrate));
int num_rtp_packets = 0;
int num_dropped_packets = 0;
int num_rtcp_packets = 0;
std::map<uint8_t, int> packet_counters;
myTransport.GetStats(num_rtp_packets, num_dropped_packets, num_rtcp_packets,
&packet_counters);
EXPECT_GT(num_rtp_packets, 0);
EXPECT_GT(num_dropped_packets, 0);
EXPECT_GT(num_rtcp_packets, 0);
EXPECT_GT(packet_counters[kRtxPayloadType], 0);
// Make sure we have lost packets and that they were retransmitted.
EXPECT_GT(recCumulativeLost, 0u);
EXPECT_GT(sentTotalBitrate, 0u);
EXPECT_GT(sentNackBitrate, 0u);
//
// Statistics
//
// Stop and restart to clear stats
ViETest::Log("Testing statistics\n");
EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, false));
EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
myTransport.ClearStats();
const int kPacketLossRate = 20;
NetworkParameters network;
network.packet_loss_rate = kPacketLossRate;
network.loss_model = kUniformLoss;
myTransport.SetNetworkParameters(network);
@ -189,16 +255,6 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
unsigned int sentExtendedMax = 0;
unsigned int sentJitter = 0;
int sentRttMs = 0;
unsigned short recFractionsLost = 0;
unsigned int recCumulativeLost = 0;
unsigned int recExtendedMax = 0;
unsigned int recJitter = 0;
int recRttMs = 0;
unsigned int sentTotalBitrate = 0;
unsigned int sentVideoBitrate = 0;
unsigned int sentFecBitrate = 0;
unsigned int sentNackBitrate = 0;
EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,

8
webrtc/video_engine/test/libvietest/include/tb_external_transport.h
View File

@ -102,9 +102,12 @@ public:
void SetSSRCFilter(uint32_t SSRC);
void ClearStats();
// |packet_counters| is a map which counts the number of packets sent per
// payload type.
void GetStats(int32_t& numRtpPackets,
int32_t& numDroppedPackets,
int32_t& numRtcpPackets);
int32_t& numRtcpPackets,
std::map<uint8_t, int>* packet_counters);
void SetTemporalToggle(unsigned char layers);
void EnableSSRCCheck();
@ -153,6 +156,9 @@ private:
int32_t _rtpCount;
int32_t _rtcpCount;
int32_t _dropCount;
// |packet_counters| is a map which counts the number of packets sent per
// payload type.
std::map<uint8_t, int> packet_counters_;
std::list<VideoPacket*> _rtpPackets;
std::list<VideoPacket*> _rtcpPackets;

8
webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc
View File

@ -53,6 +53,7 @@ TbExternalTransport::TbExternalTransport(
_rtpCount(0),
_rtcpCount(0),
_dropCount(0),
packet_counters_(),
_rtpPackets(),
_rtcpPackets(),
_send_frame_callback(NULL),
@ -109,6 +110,7 @@ int TbExternalTransport::SendPacket(int channel, const void *data, int len)
{
// Parse timestamp from RTP header according to RFC 3550, section 5.1.
uint8_t* ptr = (uint8_t*)data;
uint8_t payload_type = ptr[1] & 0x7F;
uint32_t rtp_timestamp = ptr[4] << 24;
rtp_timestamp += ptr[5] << 16;
rtp_timestamp += ptr[6] << 8;
@ -122,6 +124,7 @@ int TbExternalTransport::SendPacket(int channel, const void *data, int len)
_lastSendRTPTimestamp != rtp_timestamp) {
_send_frame_callback->FrameSent(rtp_timestamp);
}
++packet_counters_[payload_type];
_lastSendRTPTimestamp = rtp_timestamp;
if (_filterSSRC)
@ -323,16 +326,19 @@ void TbExternalTransport::ClearStats()
_rtpCount = 0;
_dropCount = 0;
_rtcpCount = 0;
packet_counters_.clear();
}
void TbExternalTransport::GetStats(int32_t& numRtpPackets,
int32_t& numDroppedPackets,
int32_t& numRtcpPackets)
int32_t& numRtcpPackets,
std::map<uint8_t, int>* packet_counters)
{
webrtc::CriticalSectionScoped cs(&_statCrit);
numRtpPackets = _rtpCount;
numDroppedPackets = _dropCount;
numRtcpPackets = _rtcpCount;
*packet_counters = packet_counters_;
}
void TbExternalTransport::EnableSSRCCheck()

8
webrtc/video_engine/vie_channel.cc
View File

@ -956,9 +956,15 @@ int32_t ViEChannel::SetSSRC(const uint32_t SSRC,
"%s(usage:%d, SSRC: 0x%x, idx:%u)",
__FUNCTION__, usage, SSRC, simulcast_idx);
if (simulcast_idx == 0) {
if (usage == kViEStreamTypeRtx) {
return rtp_rtcp_->SetRTXSendStatus(kRtxRetransmitted, true, SSRC);
}
return rtp_rtcp_->SetSSRC(SSRC);
}
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
if (rtp_rtcp_->SetSSRC(SSRC) != 0) {
return -1;
}
if (simulcast_idx > simulcast_rtp_rtcp_.size()) {
return -1;
}
@ -972,7 +978,7 @@ int32_t ViEChannel::SetSSRC(const uint32_t SSRC,
if (usage == kViEStreamTypeRtx) {
return rtp_rtcp->SetRTXSendStatus(kRtxRetransmitted, true, SSRC);
}
return rtp_rtcp->SetSSRC(SSRC);
return 0;
}
int32_t ViEChannel::SetRemoteSSRCType(const StreamType usage,

2
webrtc/video_engine/vie_rtp_rtcp_impl.cc
View File

@ -247,7 +247,7 @@ int ViERTP_RTCPImpl::SetRtxSendPayloadType(const int video_channel,
shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
return -1;
}
if (!vie_channel->SetRtxSendPayloadType(payload_type)) {
if (vie_channel->SetRtxSendPayloadType(payload_type) != 0) {
return -1;
}
return 0;