Updating NACK RTX test
BUG=1513
R=holmer@google.com, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1274006
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4036 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index b1d494c..2d1306a 100644
--- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -1,30 +1,35 @@
/*
- * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
+* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+*
+* Use of this source code is governed by a BSD-style license
+* that can be found in the LICENSE file in the root of the source
+* tree. An additional intellectual property rights grant can be found
+* in the file PATENTS. All contributing project authors may
+* be found in the AUTHORS file in the root of the source tree.
+*/
#include <algorithm>
-#include <vector>
+#include <iterator>
+#include <list>
+#include <set>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-namespace webrtc {
-const int kVideoNackListSize = 10;
+using namespace webrtc;
+
+const int kVideoNackListSize = 30;
const int kTestId = 123;
const uint32_t kTestSsrc = 3456;
const uint16_t kTestSequenceNumber = 2345;
-const uint32_t kTestNumberOfPackets = 450;
-const int kTestNumberOfRtxPackets = 49;
+const uint32_t kTestNumberOfPackets = 1350;
+const int kTestNumberOfRtxPackets = 149;
+const int kNumFrames = 30;
-class VerifyingRtxReceiver : public RtpData {
+class VerifyingRtxReceiver : public RtpData
+{
public:
VerifyingRtxReceiver() {}
@@ -32,13 +37,12 @@
const uint8_t* data,
const uint16_t size,
const webrtc::WebRtcRTPHeader* rtp_header) {
- if (!sequence_numbers_.empty()) {
+ if (!sequence_numbers_.empty())
EXPECT_EQ(kTestSsrc, rtp_header->header.ssrc);
- }
sequence_numbers_.push_back(rtp_header->header.sequenceNumber);
return 0;
}
- std::vector<uint16_t > sequence_numbers_;
+ std::list<uint16_t> sequence_numbers_;
};
class RtxLoopBackTransport : public webrtc::Transport {
@@ -46,31 +50,49 @@
explicit RtxLoopBackTransport(uint32_t rtx_ssrc)
: count_(0),
packet_loss_(0),
+ consecutive_drop_start_(0),
+ consecutive_drop_end_(0),
rtx_ssrc_(rtx_ssrc),
count_rtx_ssrc_(0),
module_(NULL) {
}
+
void SetSendModule(RtpRtcp* rtpRtcpModule) {
module_ = rtpRtcpModule;
}
+
void DropEveryNthPacket(int n) {
packet_loss_ = n;
}
+
+ void DropConsecutivePackets(int start, int total) {
+ consecutive_drop_start_ = start;
+ consecutive_drop_end_ = start + total;
+ packet_loss_ = 0;
+ }
+
virtual int SendPacket(int channel, const void *data, int len) {
count_++;
const unsigned char* ptr = static_cast<const unsigned char*>(data);
uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
if (ssrc == rtx_ssrc_) count_rtx_ssrc_++;
+ uint16_t sequence_number = (ptr[2] << 8) + ptr[3];
+ expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
+ sequence_number);
if (packet_loss_ > 0) {
if ((count_ % packet_loss_) == 0) {
return len;
}
+ } else if (count_ >= consecutive_drop_start_ &&
+ count_ < consecutive_drop_end_) {
+ return len;
}
if (module_->IncomingPacket((const uint8_t*)data, len) == 0) {
return len;
}
return -1;
}
+
virtual int SendRTCPPacket(int channel, const void *data, int len) {
if (module_->IncomingPacket((const uint8_t*)data, len) == 0) {
return len;
@@ -79,9 +101,12 @@
}
int count_;
int packet_loss_;
+ int consecutive_drop_start_;
+ int consecutive_drop_end_;
uint32_t rtx_ssrc_;
int count_rtx_ssrc_;
RtpRtcp* module_;
+ std::set<uint16_t> expected_sequence_numbers_;
};
class RtpRtcpRtxNackTest : public ::testing::Test {
@@ -126,6 +151,67 @@
}
}
+ int BuildNackList(uint16_t* nack_list) {
+ receiver_.sequence_numbers_.sort();
+ std::list<uint16_t> missing_sequence_numbers;
+ std::list<uint16_t>::iterator it =
+ receiver_.sequence_numbers_.begin();
+
+ while (it != receiver_.sequence_numbers_.end()) {
+ uint16_t sequence_number_1 = *it;
+ ++it;
+ if (it != receiver_.sequence_numbers_.end()) {
+ uint16_t sequence_number_2 = *it;
+ // Add all missing sequence numbers to list
+ for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2;
+ ++i) {
+ missing_sequence_numbers.push_back(i);
+ }
+ }
+ }
+ int n = 0;
+ for (it = missing_sequence_numbers.begin();
+ it != missing_sequence_numbers.end(); ++it) {
+ nack_list[n++] = (*it);
+ }
+ return n;
+ }
+
+ bool ExpectedPacketsReceived() {
+ std::list<uint16_t> received_sorted;
+ std::copy(receiver_.sequence_numbers_.begin(),
+ receiver_.sequence_numbers_.end(),
+ std::back_inserter(received_sorted));
+ received_sorted.sort();
+ return std::equal(received_sorted.begin(), received_sorted.end(),
+ transport_.expected_sequence_numbers_.begin());
+ }
+
+ void RunRtxTest(RtxMode rtx_method, int loss) {
+ EXPECT_EQ(0, rtp_rtcp_module_->SetRTXReceiveStatus(true, kTestSsrc + 1));
+ EXPECT_EQ(0, rtp_rtcp_module_->SetRTXSendStatus(rtx_method, true,
+ kTestSsrc + 1));
+ transport_.DropEveryNthPacket(loss);
+ uint32_t timestamp = 3000;
+ uint16_t nack_list[kVideoNackListSize];
+ for (int frame = 0; frame < kNumFrames; ++frame) {
+ EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
+ 123,
+ timestamp,
+ timestamp / 90,
+ payload_data,
+ payload_data_length));
+ int length = BuildNackList(nack_list);
+ if (length > 0)
+ rtp_rtcp_module_->SendNACK(nack_list, length);
+ fake_clock.AdvanceTimeMilliseconds(33);
+ rtp_rtcp_module_->Process();
+ // Prepare next frame.
+ timestamp += 3000;
+ }
+ receiver_.sequence_numbers_.sort();
+ }
+
virtual void TearDown() {
delete rtp_rtcp_module_;
}
@@ -138,146 +224,62 @@
SimulatedClock fake_clock;
};
-TEST_F(RtpRtcpRtxNackTest, RTCP) {
+TEST_F(RtpRtcpRtxNackTest, LongNackList) {
+ const int kNumPacketsToDrop = 900;
+ const int kNumRequiredRtcp = 4;
uint32_t timestamp = 3000;
- uint16_t nack_list[kVideoNackListSize];
- transport_.DropEveryNthPacket(10);
-
- for (int frame = 0; frame < 10; ++frame) {
+ uint16_t nack_list[kNumPacketsToDrop];
+ // Disable StorePackets to be able to set a larger packet history.
+ EXPECT_EQ(0, rtp_rtcp_module_->SetStorePacketsStatus(false, 0));
+ // Enable StorePackets with a packet history of 2000 packets.
+ EXPECT_EQ(0, rtp_rtcp_module_->SetStorePacketsStatus(true, 2000));
+ // Drop 900 packets from the second one so that we get a NACK list which is
+ // big enough to require 4 RTCP packets to be fully transmitted to the sender.
+ transport_.DropConsecutivePackets(2, kNumPacketsToDrop);
+ // Send 30 frames which at the default size is roughly what we need to get
+ // enough packets.
+ for (int frame = 0; frame < kNumFrames; ++frame) {
EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
123,
timestamp,
timestamp / 90,
payload_data,
payload_data_length));
-
- std::sort(receiver_.sequence_numbers_.begin(),
- receiver_.sequence_numbers_.end());
-
- std::vector<uint16_t> missing_sequence_numbers;
- std::vector<uint16_t>::iterator it =
- receiver_.sequence_numbers_.begin();
-
- while (it != receiver_.sequence_numbers_.end()) {
- uint16_t sequence_number_1 = *it;
- ++it;
- if (it != receiver_.sequence_numbers_.end()) {
- uint16_t sequence_number_2 = *it;
- // Add all missing sequence numbers to list.
- for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2;
- ++i) {
- missing_sequence_numbers.push_back(i);
- }
- }
- }
- int n = 0;
- for (it = missing_sequence_numbers.begin();
- it != missing_sequence_numbers.end(); ++it) {
- nack_list[n++] = (*it);
- }
- rtp_rtcp_module_->SendNACK(nack_list, n);
- fake_clock.AdvanceTimeMilliseconds(33);
- rtp_rtcp_module_->Process();
-
// Prepare next frame.
timestamp += 3000;
+ fake_clock.AdvanceTimeMilliseconds(33);
+ rtp_rtcp_module_->Process();
}
- std::sort(receiver_.sequence_numbers_.begin(),
- receiver_.sequence_numbers_.end());
- EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
- EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
- *(receiver_.sequence_numbers_.rbegin()));
- EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
- EXPECT_EQ(0, transport_.count_rtx_ssrc_);
+ EXPECT_FALSE(transport_.expected_sequence_numbers_.empty());
+ EXPECT_FALSE(receiver_.sequence_numbers_.empty());
+ size_t last_receive_count = receiver_.sequence_numbers_.size();
+ int length = BuildNackList(nack_list);
+ for (int i = 0; i < kNumRequiredRtcp - 1; ++i) {
+ rtp_rtcp_module_->SendNACK(nack_list, length);
+ EXPECT_GT(receiver_.sequence_numbers_.size(), last_receive_count);
+ last_receive_count = receiver_.sequence_numbers_.size();
+ EXPECT_FALSE(ExpectedPacketsReceived());
+ }
+ rtp_rtcp_module_->SendNACK(nack_list, length);
+ EXPECT_GT(receiver_.sequence_numbers_.size(), last_receive_count);
+ EXPECT_TRUE(ExpectedPacketsReceived());
}
-TEST_F(RtpRtcpRtxNackTest, RTXNack) {
- EXPECT_EQ(0, rtp_rtcp_module_->SetRTXReceiveStatus(true, kTestSsrc + 1));
- rtp_rtcp_module_->SetRtxReceivePayloadType(119);
- EXPECT_EQ(0, rtp_rtcp_module_->SetRTXSendStatus(kRtxRetransmitted, true,
- kTestSsrc + 1));
- rtp_rtcp_module_->SetRtxSendPayloadType(119);
-
- transport_.DropEveryNthPacket(10);
-
- uint32_t timestamp = 3000;
- uint16_t nack_list[kVideoNackListSize];
-
- for (int frame = 0; frame < 10; ++frame) {
- EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
- 123,
- timestamp,
- timestamp / 90,
- payload_data,
- payload_data_length));
-
- std::sort(receiver_.sequence_numbers_.begin(),
- receiver_.sequence_numbers_.end());
-
- std::vector<uint16_t> missing_sequence_numbers;
-
-
- std::vector<uint16_t>::iterator it =
- receiver_.sequence_numbers_.begin();
- while (it != receiver_.sequence_numbers_.end()) {
- int sequence_number_1 = *it;
- ++it;
- if (it != receiver_.sequence_numbers_.end()) {
- int sequence_number_2 = *it;
- // Add all missing sequence numbers to list.
- for (int i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
- missing_sequence_numbers.push_back(i);
- }
- }
- }
- int n = 0;
- for (it = missing_sequence_numbers.begin();
- it != missing_sequence_numbers.end(); ++it) {
- nack_list[n++] = (*it);
- }
- rtp_rtcp_module_->SendNACK(nack_list, n);
- fake_clock.AdvanceTimeMilliseconds(33);
- rtp_rtcp_module_->Process();
-
- // Prepare next frame.
- timestamp += 3000;
- }
- std::sort(receiver_.sequence_numbers_.begin(),
- receiver_.sequence_numbers_.end());
+TEST_F(RtpRtcpRtxNackTest, RtxNack) {
+ RunRtxTest(kRtxRetransmitted, 10);
EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
- *(receiver_.sequence_numbers_.rbegin()));
+ *(receiver_.sequence_numbers_.rbegin()));
EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
+ EXPECT_TRUE(ExpectedPacketsReceived());
}
TEST_F(RtpRtcpRtxNackTest, RTXAllNoLoss) {
- EXPECT_EQ(0, rtp_rtcp_module_->SetRTXReceiveStatus(true, kTestSsrc + 1));
- EXPECT_EQ(0, rtp_rtcp_module_->SetRTXSendStatus(kRtxAll, true,
- kTestSsrc + 1));
- transport_.DropEveryNthPacket(0);
-
- uint32_t timestamp = 3000;
-
- for (int frame = 0; frame < 10; ++frame) {
- EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
- 123,
- timestamp,
- timestamp / 90,
- payload_data,
- payload_data_length));
-
- fake_clock.AdvanceTimeMilliseconds(33);
- rtp_rtcp_module_->Process();
-
- // Prepare next frame.
- timestamp += 3000;
- }
- std::sort(receiver_.sequence_numbers_.begin(),
- receiver_.sequence_numbers_.end());
+ RunRtxTest(kRtxAll, 0);
EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
- *(receiver_.sequence_numbers_.rbegin()));
+ *(receiver_.sequence_numbers_.rbegin()));
// We have transmitted all packets twice, and loss was set to 0.
EXPECT_EQ(kTestNumberOfPackets * 2u, receiver_.sequence_numbers_.size());
// Half of the packets should be via RTX.
@@ -286,63 +288,13 @@
}
TEST_F(RtpRtcpRtxNackTest, RTXAllWithLoss) {
- EXPECT_EQ(0, rtp_rtcp_module_->SetRTXReceiveStatus(true, kTestSsrc + 1));
- EXPECT_EQ(0, rtp_rtcp_module_->SetRTXSendStatus(kRtxAll, true,
- kTestSsrc + 1));
-
int loss = 10;
- transport_.DropEveryNthPacket(loss);
-
- uint32_t timestamp = 3000;
- uint16_t nack_list[kVideoNackListSize];
-
- for (int frame = 0; frame < 10; ++frame) {
- EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
- 123,
- timestamp,
- timestamp / 90,
- payload_data,
- payload_data_length));
- std::sort(receiver_.sequence_numbers_.begin(),
- receiver_.sequence_numbers_.end());
- std::vector<uint16_t> missing_sequence_numbers;
-
- std::vector<uint16_t>::iterator it =
- receiver_.sequence_numbers_.begin();
- while (it != receiver_.sequence_numbers_.end()) {
- int sequence_number_1 = *it;
- ++it;
- if (it != receiver_.sequence_numbers_.end()) {
- int sequence_number_2 = *it;
- // Add all missing sequence numbers to list.
- for (int i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
- missing_sequence_numbers.push_back(i);
- }
- }
- }
- int n = 0;
- for (it = missing_sequence_numbers.begin();
- it != missing_sequence_numbers.end(); ++it) {
- nack_list[n++] = (*it);
- }
- if (n > 0)
- rtp_rtcp_module_->SendNACK(nack_list, n);
- fake_clock.AdvanceTimeMilliseconds(33);
- rtp_rtcp_module_->Process();
-
- // Prepare next frame.
- timestamp += 3000;
- }
- std::sort(receiver_.sequence_numbers_.begin(),
- receiver_.sequence_numbers_.end());
+ RunRtxTest(kRtxAll, loss);
EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
- *(receiver_.sequence_numbers_.rbegin()));
- // Got everything but 10% loss.
- EXPECT_EQ(2u * (kTestNumberOfPackets - kTestNumberOfPackets / 10),
- receiver_.sequence_numbers_.size());
- EXPECT_EQ(static_cast<int>(kTestNumberOfPackets),
- transport_.count_rtx_ssrc_);
+ *(receiver_.sequence_numbers_.rbegin()));
+ // Got everything but lost packets.
+ EXPECT_EQ(2u * (kTestNumberOfPackets - kTestNumberOfPackets / loss),
+ receiver_.sequence_numbers_.size());
+ EXPECT_EQ(static_cast<int>(kTestNumberOfPackets), transport_.count_rtx_ssrc_);
}
-
-} // namespace webrtc
