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Unified Diff: webrtc/modules/pacing/packet_router_unittest.cc

Issue 1247293002: Add support for transport wide sequence numbers (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 5 months ago
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« webrtc/modules/pacing/packet_router.cc ('K') | « webrtc/modules/pacing/packet_router.cc ('k') | webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender.cc » ('J')
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Index: webrtc/modules/pacing/packet_router_unittest.cc
diff --git a/webrtc/modules/pacing/packet_router_unittest.cc b/webrtc/modules/pacing/packet_router_unittest.cc
index f7fdf7bbcaee84905669a4fb66421fada55a338f..ef589641c191bc007c21e176f748ae94e4df5a11 100644
--- a/webrtc/modules/pacing/packet_router_unittest.cc
+++ b/webrtc/modules/pacing/packet_router_unittest.cc
@@ -33,19 +33,22 @@ class PacketRouterTest : public ::testing::Test {
};
TEST_F(PacketRouterTest, TimeToSendPacket) {
+ const uint16_t kSsrc1 = 1234;
+ const uint16_t kSsrc2 = 4567;
+
MockRtpRtcp rtp_1;
+ EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
MockRtpRtcp rtp_2;
+ EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
packet_router_->AddRtpModule(&rtp_1);
packet_router_->AddRtpModule(&rtp_2);
- const uint16_t kSsrc1 = 1234;
uint16_t sequence_number = 17;
uint64_t timestamp = 7890;
bool retransmission = false;
// Send on the first module by letting rtp_1 be sending with correct ssrc.
EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
- EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc1, sequence_number, timestamp,
retransmission))
.Times(1)
@@ -58,15 +61,15 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
++sequence_number;
timestamp += 30;
retransmission = true;
- const uint16_t kSsrc2 = 4567;
EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
- EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
EXPECT_CALL(rtp_2, TimeToSendPacket(kSsrc2, sequence_number, timestamp,
retransmission))
.Times(1)
.WillOnce(Return(true));
+ EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
+ timestamp, retransmission));
EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc2, sequence_number,
timestamp, retransmission));
@@ -77,12 +80,10 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
timestamp, retransmission));
+ EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc2, sequence_number,
+ timestamp, retransmission));
// Add a packet with incorrect ssrc and test it's dropped in the router.
- EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
- EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
- EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
- EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1 + kSsrc2, sequence_number,
@@ -92,8 +93,6 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
// rtp_1 has been removed, try sending a packet on that ssrc and make sure
// it is dropped as expected by not expecting any calls to rtp_1.
- EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
- EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
timestamp, retransmission));
@@ -102,20 +101,30 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
}
TEST_F(PacketRouterTest, TimeToSendPadding) {
+ const uint16_t kSsrc1 = 1234;
+ const uint16_t kSsrc2 = 4567;
+
MockRtpRtcp rtp_1;
+ EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
MockRtpRtcp rtp_2;
+ EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
packet_router_->AddRtpModule(&rtp_1);
packet_router_->AddRtpModule(&rtp_2);
- // Default configuration, sending padding on the first sending module.
+ // Default configuration, sending padding on all modules sending media,
+ // ordered by SSRC.
const size_t requested_padding_bytes = 1000;
const size_t sent_padding_bytes = 890;
EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
.Times(1)
.WillOnce(Return(sent_padding_bytes));
- EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0);
- EXPECT_EQ(sent_padding_bytes,
+ EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
+ EXPECT_CALL(rtp_2,
+ TimeToSendPadding(requested_padding_bytes - sent_padding_bytes))
+ .Times(1)
+ .WillOnce(Return(requested_padding_bytes - sent_padding_bytes));
+ EXPECT_EQ(requested_padding_bytes,
packet_router_->TimeToSendPadding(requested_padding_bytes));
// Let only the second module be sending and verify the padding request is
@@ -134,8 +143,7 @@ TEST_F(PacketRouterTest, TimeToSendPadding) {
EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0);
EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0);
- EXPECT_EQ(static_cast<size_t>(0),
- packet_router_->TimeToSendPadding(requested_padding_bytes));
+ EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
packet_router_->RemoveRtpModule(&rtp_1);
@@ -143,9 +151,108 @@ TEST_F(PacketRouterTest, TimeToSendPadding) {
// to send by not expecting any calls. Instead verify rtp_2 is called.
EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes)).Times(1);
- EXPECT_EQ(static_cast<size_t>(0),
- packet_router_->TimeToSendPadding(requested_padding_bytes));
+ EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
packet_router_->RemoveRtpModule(&rtp_2);
}
+
+TEST_F(PacketRouterTest, ChangingSsrc) {
+ const uint16_t kSsrc1 = 1234;
+ const uint16_t kSsrc2 = 4567;
+ uint16_t sequence_number = 17;
+ uint64_t timestamp = 7890;
+ bool retransmission = false;
+
+ MockRtpRtcp rtp_1;
+ EXPECT_CALL(rtp_1, SendingMedia()).WillRepeatedly(Return(true));
+
+ EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
+ packet_router_->AddRtpModule(&rtp_1);
+
+ EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc1, sequence_number, timestamp,
+ retransmission))
+ .Times(1)
+ .WillOnce(Return(true));
+ EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
+ timestamp, retransmission));
+
+ packet_router_->OnSsrcChanged();
+
+ EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc2));
+ EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc2, sequence_number, timestamp,
+ retransmission))
+ .Times(1)
+ .WillOnce(Return(true));
+ EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc2, sequence_number,
+ timestamp, retransmission));
+
+ EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc2));
+ packet_router_->RemoveRtpModule(&rtp_1);
+}
+
+TEST_F(PacketRouterTest, TransportSeqNotEnabled) {
+ packet_router_->SetTransportWideSequenceNumber(1000);
+ EXPECT_EQ(0u, packet_router_->AllocateSequenceNumber());
+
+ const uint16_t kSeq = 1;
+ packet_router_->OnPacketSent(kSeq, 1);
+ std::vector<PacketInfo> packet_vec;
+ packet_vec.push_back(PacketInfo(0, 0, kSeq, 0, false));
+ EXPECT_EQ(0u, packet_router_->PopulateSendTimes(&packet_vec));
+}
+
+TEST_F(PacketRouterTest, AddAndPopulateTransportSeq) {
+ packet_router_->EnableTransportWideFeedback();
+
+ const uint16_t kStartSeq = 0xFFF0;
+ const int64_t kStartTime = 2000;
+ const size_t kNumPackets = 32;
+
+ packet_router_->SetTransportWideSequenceNumber(kStartSeq - 1);
+ std::vector<PacketInfo> packet_vec;
+
+ for (size_t i = 0; i < kNumPackets; ++i) {
+ uint16_t seq = packet_router_->AllocateSequenceNumber();
+ packet_router_->OnPacketSent(seq, kStartTime + i);
+ packet_vec.push_back(PacketInfo(0, 0, seq, 0, false));
+ }
+
+ EXPECT_EQ(kNumPackets, packet_router_->PopulateSendTimes(&packet_vec));
+ for (size_t i = 0; i < kNumPackets; ++i) {
+ EXPECT_EQ(static_cast<uint16_t>(kStartSeq + i),
+ packet_vec[i].sequence_number);
+ EXPECT_EQ(static_cast<int64_t>(kStartTime + i), packet_vec[i].send_time_ms);
+ }
+}
+
+TEST_F(PacketRouterTest, AddAndPopulateTransportSeqWithLosses) {
+ packet_router_->EnableTransportWideFeedback();
+
+ const uint16_t kStartSeq = 0xFFF0;
+ const int64_t kStartTime = 2000;
+ const size_t kNumPackets = 32;
+
+ packet_router_->SetTransportWideSequenceNumber(kStartSeq - 1);
+ std::vector<PacketInfo> packet_vec;
+
+ for (size_t i = 0; i < kNumPackets; ++i) {
+ uint16_t seq = packet_router_->AllocateSequenceNumber();
+ packet_router_->OnPacketSent(seq, kStartTime + i);
+ packet_vec.push_back(PacketInfo(0, 0, seq, 0, false));
+ }
+
+ const size_t kNumReceivedPackets = 16;
+ const size_t kFirstLossPos = 5;
+ packet_vec.erase(
+ packet_vec.begin() + kFirstLossPos,
+ packet_vec.begin() + kFirstLossPos + (kNumPackets - kNumReceivedPackets));
+
+ EXPECT_EQ(kNumReceivedPackets,
+ packet_router_->PopulateSendTimes(&packet_vec));
+ for (const PacketInfo& info : packet_vec) {
+ uint16_t seq_delta = info.sequence_number - kStartSeq;
+ EXPECT_EQ(static_cast<int64_t>(kStartTime + seq_delta), info.send_time_ms);
+ }
+}
+
} // namespace webrtc
« webrtc/modules/pacing/packet_router.cc ('K') | « webrtc/modules/pacing/packet_router.cc ('k') | webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender.cc » ('J')

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