webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
tnakamura@webrtc.org aa4d96a134 Revert r4301
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00

432 lines
15 KiB
C++

/*
* Copyright (c) 2012 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.
*/
/*
* This file includes unit tests for the RTCPReceiver.
*/
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
// Note: This file has no directory. Lint warning must be ignored.
#include "webrtc/common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
namespace webrtc {
namespace { // Anonymous namespace; hide utility functions and classes.
// A very simple packet builder class for building RTCP packets.
class PacketBuilder {
public:
static const int kMaxPacketSize = 1024;
PacketBuilder()
: pos_(0),
pos_of_len_(0) {
}
void Add8(uint8_t byte) {
EXPECT_LT(pos_, kMaxPacketSize - 1);
buffer_[pos_] = byte;
++ pos_;
}
void Add16(uint16_t word) {
Add8(word >> 8);
Add8(word & 0xFF);
}
void Add32(uint32_t word) {
Add8(word >> 24);
Add8((word >> 16) & 0xFF);
Add8((word >> 8) & 0xFF);
Add8(word & 0xFF);
}
void Add64(uint32_t upper_half, uint32_t lower_half) {
Add32(upper_half);
Add32(lower_half);
}
// Set the 5-bit value in the 1st byte of the header
// and the payload type. Set aside room for the length field,
// and make provision for backpatching it.
// Note: No way to set the padding bit.
void AddRtcpHeader(int payload, int format_or_count) {
PatchLengthField();
Add8(0x80 | (format_or_count & 0x1F));
Add8(payload);
pos_of_len_ = pos_;
Add16(0xDEAD); // Initialize length to "clearly illegal".
}
void AddTmmbrBandwidth(int mantissa, int exponent, int overhead) {
// 6 bits exponent, 17 bits mantissa, 9 bits overhead.
uint32_t word = 0;
word |= (exponent << 26);
word |= ((mantissa & 0x1FFFF) << 9);
word |= (overhead & 0x1FF);
Add32(word);
}
void AddSrPacket(uint32_t sender_ssrc) {
AddRtcpHeader(200, 0);
Add32(sender_ssrc);
Add64(0x10203, 0x4050607); // NTP timestamp
Add32(0x10203); // RTP timestamp
Add32(0); // Sender's packet count
Add32(0); // Sender's octet count
}
void AddRrPacket(uint32_t sender_ssrc, uint32_t rtp_ssrc,
uint32_t extended_max) {
AddRtcpHeader(201, 1);
Add32(sender_ssrc);
Add32(rtp_ssrc);
Add32(0); // No loss.
Add32(extended_max);
Add32(0); // Jitter.
Add32(0); // Last SR.
Add32(0); // Delay since last SR.
}
const uint8_t* packet() {
PatchLengthField();
return buffer_;
}
unsigned int length() {
return pos_;
}
private:
void PatchLengthField() {
if (pos_of_len_ > 0) {
// Backpatch the packet length. The client must have taken
// care of proper padding to 32-bit words.
int this_packet_length = (pos_ - pos_of_len_ - 2);
ASSERT_EQ(0, this_packet_length % 4)
<< "Packets must be a multiple of 32 bits long"
<< " pos " << pos_ << " pos_of_len " << pos_of_len_;
buffer_[pos_of_len_] = this_packet_length >> 10;
buffer_[pos_of_len_+1] = (this_packet_length >> 2) & 0xFF;
pos_of_len_ = 0;
}
}
int pos_;
// Where the length field of the current packet is.
// Note that 0 is not a legal value, so is used for "uninitialized".
int pos_of_len_;
uint8_t buffer_[kMaxPacketSize];
};
// This test transport verifies that no functions get called.
class TestTransport : public Transport,
public RtpData {
public:
explicit TestTransport()
: rtcp_receiver_(NULL) {
}
void SetRTCPReceiver(RTCPReceiver* rtcp_receiver) {
rtcp_receiver_ = rtcp_receiver;
}
virtual int SendPacket(int /*ch*/, const void* /*data*/, int /*len*/) {
ADD_FAILURE(); // FAIL() gives a compile error.
return -1;
}
// Injects an RTCP packet into the receiver.
virtual int SendRTCPPacket(int /* ch */, const void *packet, int packet_len) {
ADD_FAILURE();
return 0;
}
virtual int OnReceivedPayloadData(const uint8_t* payloadData,
const uint16_t payloadSize,
const WebRtcRTPHeader* rtpHeader) {
ADD_FAILURE();
return 0;
}
RTCPReceiver* rtcp_receiver_;
};
class RtcpReceiverTest : public ::testing::Test {
protected:
RtcpReceiverTest()
: over_use_detector_options_(),
system_clock_(1335900000),
remote_bitrate_observer_(),
remote_bitrate_estimator_(
RemoteBitrateEstimatorFactory().Create(
&remote_bitrate_observer_,
&system_clock_)) {
test_transport_ = new TestTransport();
RtpRtcp::Configuration configuration;
configuration.id = 0;
configuration.audio = false;
configuration.clock = &system_clock_;
configuration.outgoing_transport = test_transport_;
configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
rtcp_receiver_ = new RTCPReceiver(0, &system_clock_, rtp_rtcp_impl_);
test_transport_->SetRTCPReceiver(rtcp_receiver_);
}
~RtcpReceiverTest() {
delete rtcp_receiver_;
delete rtp_rtcp_impl_;
delete test_transport_;
}
// Injects an RTCP packet into the receiver.
// Returns 0 for OK, non-0 for failure.
int InjectRtcpPacket(const uint8_t* packet,
uint16_t packet_len) {
RTCPUtility::RTCPParserV2 rtcpParser(packet,
packet_len,
true); // Allow non-compound RTCP
RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
int result = rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
&rtcpParser);
// The NACK list is on purpose not copied below as it isn't needed by the
// test.
rtcp_packet_info_.rtcpPacketTypeFlags =
rtcpPacketInformation.rtcpPacketTypeFlags;
rtcp_packet_info_.remoteSSRC = rtcpPacketInformation.remoteSSRC;
rtcp_packet_info_.applicationSubType =
rtcpPacketInformation.applicationSubType;
rtcp_packet_info_.applicationName = rtcpPacketInformation.applicationName;
rtcp_packet_info_.reportBlock = rtcpPacketInformation.reportBlock;
rtcp_packet_info_.fractionLost = rtcpPacketInformation.fractionLost;
rtcp_packet_info_.roundTripTime = rtcpPacketInformation.roundTripTime;
rtcp_packet_info_.lastReceivedExtendedHighSeqNum =
rtcpPacketInformation.lastReceivedExtendedHighSeqNum;
rtcp_packet_info_.jitter = rtcpPacketInformation.jitter;
rtcp_packet_info_.interArrivalJitter =
rtcpPacketInformation.interArrivalJitter;
rtcp_packet_info_.sliPictureId = rtcpPacketInformation.sliPictureId;
rtcp_packet_info_.rpsiPictureId = rtcpPacketInformation.rpsiPictureId;
rtcp_packet_info_.receiverEstimatedMaxBitrate =
rtcpPacketInformation.receiverEstimatedMaxBitrate;
rtcp_packet_info_.ntp_secs = rtcpPacketInformation.ntp_secs;
rtcp_packet_info_.ntp_frac = rtcpPacketInformation.ntp_frac;
rtcp_packet_info_.rtp_timestamp = rtcpPacketInformation.rtp_timestamp;
return result;
}
OverUseDetectorOptions over_use_detector_options_;
SimulatedClock system_clock_;
ModuleRtpRtcpImpl* rtp_rtcp_impl_;
RTCPReceiver* rtcp_receiver_;
TestTransport* test_transport_;
RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
MockRemoteBitrateObserver remote_bitrate_observer_;
scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
};
TEST_F(RtcpReceiverTest, BrokenPacketIsIgnored) {
const uint8_t bad_packet[] = {0, 0, 0, 0};
EXPECT_EQ(0, InjectRtcpPacket(bad_packet, sizeof(bad_packet)));
EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
}
TEST_F(RtcpReceiverTest, InjectSrPacket) {
const uint32_t kSenderSsrc = 0x10203;
PacketBuilder p;
p.AddSrPacket(kSenderSsrc);
EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
// The parser will note the remote SSRC on a SR from other than his
// expected peer, but will not flag that he's gotten a packet.
EXPECT_EQ(kSenderSsrc, rtcp_packet_info_.remoteSSRC);
EXPECT_EQ(0U,
kRtcpSr & rtcp_packet_info_.rtcpPacketTypeFlags);
}
TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
const uint32_t kSenderSsrc = 0x10203;
const uint32_t kSourceSsrc = 0x40506;
const int64_t kRtcpIntervalMs = 1000;
rtcp_receiver_->SetSSRC(kSourceSsrc);
uint32_t sequence_number = 1234;
system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
// No RR received, shouldn't trigger a timeout.
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
// Add a RR and advance the clock just enough to not trigger a timeout.
PacketBuilder p1;
p1.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
EXPECT_EQ(0, InjectRtcpPacket(p1.packet(), p1.length()));
system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs - 1);
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
// Add a RR with the same extended max as the previous RR to trigger a
// sequence number timeout, but not a RR timeout.
PacketBuilder p2;
p2.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
EXPECT_EQ(0, InjectRtcpPacket(p2.packet(), p2.length()));
system_clock_.AdvanceTimeMilliseconds(2);
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
EXPECT_TRUE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
// Advance clock enough to trigger an RR timeout too.
system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
EXPECT_TRUE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
// We should only get one timeout even though we still haven't received a new
// RR.
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
// Add a new RR with increase sequence number to reset timers.
PacketBuilder p3;
sequence_number++;
p2.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
EXPECT_EQ(0, InjectRtcpPacket(p2.packet(), p2.length()));
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
// Verify we can get a timeout again once we've received new RR.
system_clock_.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
PacketBuilder p4;
p4.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
EXPECT_EQ(0, InjectRtcpPacket(p4.packet(), p4.length()));
system_clock_.AdvanceTimeMilliseconds(kRtcpIntervalMs + 1);
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
EXPECT_TRUE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
system_clock_.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
EXPECT_TRUE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
}
TEST_F(RtcpReceiverTest, TmmbrReceivedWithNoIncomingPacket) {
// This call is expected to fail because no data has arrived.
EXPECT_EQ(-1, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
}
TEST_F(RtcpReceiverTest, TmmbrPacketAccepted) {
const uint32_t kMediaFlowSsrc = 0x2040608;
const uint32_t kSenderSsrc = 0x10203;
const uint32_t kMediaRecipientSsrc = 0x101;
rtcp_receiver_->SetSSRC(kMediaFlowSsrc); // Matches "media source" above.
PacketBuilder p;
p.AddSrPacket(kSenderSsrc);
// TMMBR packet.
p.AddRtcpHeader(205, 3);
p.Add32(kSenderSsrc);
p.Add32(kMediaRecipientSsrc);
p.Add32(kMediaFlowSsrc);
p.AddTmmbrBandwidth(30000, 0, 0); // 30 Kbits/sec bandwidth, no overhead.
EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
TMMBRSet candidate_set;
candidate_set.VerifyAndAllocateSet(1);
EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(1, 0, &candidate_set));
EXPECT_LT(0U, candidate_set.Tmmbr(0));
EXPECT_EQ(kMediaRecipientSsrc, candidate_set.Ssrc(0));
}
TEST_F(RtcpReceiverTest, TmmbrPacketNotForUsIgnored) {
const uint32_t kMediaFlowSsrc = 0x2040608;
const uint32_t kSenderSsrc = 0x10203;
const uint32_t kMediaRecipientSsrc = 0x101;
const uint32_t kOtherMediaFlowSsrc = 0x9999;
PacketBuilder p;
p.AddSrPacket(kSenderSsrc);
// TMMBR packet.
p.AddRtcpHeader(205, 3);
p.Add32(kSenderSsrc);
p.Add32(kMediaRecipientSsrc);
p.Add32(kOtherMediaFlowSsrc); // This SSRC is not what we're sending.
p.AddTmmbrBandwidth(30000, 0, 0);
rtcp_receiver_->SetSSRC(kMediaFlowSsrc);
EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
}
TEST_F(RtcpReceiverTest, TmmbrPacketZeroRateIgnored) {
const uint32_t kMediaFlowSsrc = 0x2040608;
const uint32_t kSenderSsrc = 0x10203;
const uint32_t kMediaRecipientSsrc = 0x101;
rtcp_receiver_->SetSSRC(kMediaFlowSsrc); // Matches "media source" above.
PacketBuilder p;
p.AddSrPacket(kSenderSsrc);
// TMMBR packet.
p.AddRtcpHeader(205, 3);
p.Add32(kSenderSsrc);
p.Add32(kMediaRecipientSsrc);
p.Add32(kMediaFlowSsrc);
p.AddTmmbrBandwidth(0, 0, 0); // Rate zero.
EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
}
TEST_F(RtcpReceiverTest, TmmbrThreeConstraintsTimeOut) {
const uint32_t kMediaFlowSsrc = 0x2040608;
const uint32_t kSenderSsrc = 0x10203;
const uint32_t kMediaRecipientSsrc = 0x101;
rtcp_receiver_->SetSSRC(kMediaFlowSsrc); // Matches "media source" above.
// Inject 3 packets "from" kMediaRecipientSsrc, Ssrc+1, Ssrc+2.
// The times of arrival are starttime + 0, starttime + 5 and starttime + 10.
for (uint32_t ssrc = kMediaRecipientSsrc;
ssrc < kMediaRecipientSsrc+3; ++ssrc) {
PacketBuilder p;
p.AddSrPacket(kSenderSsrc);
// TMMBR packet.
p.AddRtcpHeader(205, 3);
p.Add32(kSenderSsrc);
p.Add32(ssrc);
p.Add32(kMediaFlowSsrc);
p.AddTmmbrBandwidth(30000, 0, 0); // 30 Kbits/sec bandwidth, no overhead.
EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
// 5 seconds between each packet.
system_clock_.AdvanceTimeMilliseconds(5000);
}
// It is now starttime+15.
EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
TMMBRSet candidate_set;
candidate_set.VerifyAndAllocateSet(3);
EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(3, 0, &candidate_set));
EXPECT_LT(0U, candidate_set.Tmmbr(0));
// We expect the timeout to be 25 seconds. Advance the clock by 12
// seconds, timing out the first packet.
system_clock_.AdvanceTimeMilliseconds(12000);
// Odd behaviour: Just counting them does not trigger the timeout.
EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
// Odd behaviour: There's only one left after timeout, not 2.
EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(3, 0, &candidate_set));
EXPECT_EQ(kMediaRecipientSsrc + 2, candidate_set.Ssrc(0));
}
} // Anonymous namespace
} // namespace webrtc