webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
Jiawei Ou 8b5d9d8650 Remove the audio/video split for the RTCP report intervals.
This is a follow up of a comment in
https://webrtc-review.googlesource.com/c/src/+/110105

It was not very useful to split the audio and video report interval since the RTCP module can only either be audio or video.

The recent it was written that way in https://webrtc-review.googlesource.com/c/src/+/43201/ was because that was a straightforward transition from two global constants to two variable.

Bug: webrtc:8789
Change-Id: I2293de14ba5f363351f379a02022ed5dc7b8d458
Reviewed-on: https://webrtc-review.googlesource.com/c/110824
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25741}
2018-11-22 01:39:41 +00:00

751 lines
31 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.
*/
#include <memory>
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "rtc_base/rate_limiter.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_transport.h"
#include "test/rtcp_packet_parser.h"
using ::testing::_;
using ::testing::ElementsAre;
using ::testing::Invoke;
using ::testing::SizeIs;
namespace webrtc {
class RtcpPacketTypeCounterObserverImpl : public RtcpPacketTypeCounterObserver {
public:
RtcpPacketTypeCounterObserverImpl() : ssrc_(0) {}
~RtcpPacketTypeCounterObserverImpl() override = default;
void RtcpPacketTypesCounterUpdated(
uint32_t ssrc,
const RtcpPacketTypeCounter& packet_counter) override {
ssrc_ = ssrc;
counter_ = packet_counter;
}
uint32_t ssrc_;
RtcpPacketTypeCounter counter_;
};
class TestTransport : public Transport {
public:
TestTransport() {}
bool SendRtp(const uint8_t* /*data*/,
size_t /*len*/,
const PacketOptions& options) override {
return false;
}
bool SendRtcp(const uint8_t* data, size_t len) override {
parser_.Parse(data, len);
return true;
}
test::RtcpPacketParser parser_;
};
namespace {
static const uint32_t kSenderSsrc = 0x11111111;
static const uint32_t kRemoteSsrc = 0x22222222;
static const uint32_t kStartRtpTimestamp = 0x34567;
static const uint32_t kRtpTimestamp = 0x45678;
} // namespace
class RtcpSenderTest : public ::testing::Test {
protected:
RtcpSenderTest()
: clock_(1335900000),
receive_statistics_(ReceiveStatistics::Create(&clock_)),
retransmission_rate_limiter_(&clock_, 1000) {
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.clock = &clock_;
configuration.outgoing_transport = &test_transport_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
configuration.rtcp_report_interval_ms = 1000;
rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
nullptr, nullptr, &test_transport_,
configuration.rtcp_report_interval_ms));
rtcp_sender_->SetSSRC(kSenderSsrc);
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds(),
/*paylpad_type=*/0);
}
void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
RtpPacketReceived packet;
packet.SetSsrc(remote_ssrc);
packet.SetSequenceNumber(seq_num);
packet.SetTimestamp(12345);
packet.SetPayloadSize(100 - 12);
receive_statistics_->OnRtpPacket(packet);
}
test::RtcpPacketParser* parser() { return &test_transport_.parser_; }
RTCPSender::FeedbackState feedback_state() {
return rtp_rtcp_impl_->GetFeedbackState();
}
SimulatedClock clock_;
TestTransport test_transport_;
std::unique_ptr<ReceiveStatistics> receive_statistics_;
std::unique_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
std::unique_ptr<RTCPSender> rtcp_sender_;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtcpSenderTest, SetRtcpStatus) {
EXPECT_EQ(RtcpMode::kOff, rtcp_sender_->Status());
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(RtcpMode::kReducedSize, rtcp_sender_->Status());
}
TEST_F(RtcpSenderTest, SetSendingStatus) {
EXPECT_FALSE(rtcp_sender_->Sending());
EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), true));
EXPECT_TRUE(rtcp_sender_->Sending());
}
TEST_F(RtcpSenderTest, NoPacketSentIfOff) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kOff);
EXPECT_EQ(-1, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSr));
}
TEST_F(RtcpSenderTest, SendSr) {
const uint32_t kPacketCount = 0x12345;
const uint32_t kOctetCount = 0x23456;
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
rtcp_sender_->SetSendingStatus(feedback_state, true);
feedback_state.packets_sent = kPacketCount;
feedback_state.media_bytes_sent = kOctetCount;
NtpTime ntp = TimeMicrosToNtp(clock_.TimeInMicroseconds());
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr));
EXPECT_EQ(1, parser()->sender_report()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->sender_report()->sender_ssrc());
EXPECT_EQ(ntp, parser()->sender_report()->ntp());
EXPECT_EQ(kPacketCount, parser()->sender_report()->sender_packet_count());
EXPECT_EQ(kOctetCount, parser()->sender_report()->sender_octet_count());
EXPECT_EQ(kStartRtpTimestamp + kRtpTimestamp,
parser()->sender_report()->rtp_timestamp());
EXPECT_EQ(0U, parser()->sender_report()->report_blocks().size());
}
TEST_F(RtcpSenderTest, SendConsecutiveSrWithExactSlope) {
const uint32_t kPacketCount = 0x12345;
const uint32_t kOctetCount = 0x23456;
const int kTimeBetweenSRsUs = 10043; // Not exact value in milliseconds.
const int kExtraPackets = 30;
// Make sure clock is not exactly at some milliseconds point.
clock_.AdvanceTimeMicroseconds(kTimeBetweenSRsUs);
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
rtcp_sender_->SetSendingStatus(feedback_state, true);
feedback_state.packets_sent = kPacketCount;
feedback_state.media_bytes_sent = kOctetCount;
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr));
EXPECT_EQ(1, parser()->sender_report()->num_packets());
NtpTime ntp1 = parser()->sender_report()->ntp();
uint32_t rtp1 = parser()->sender_report()->rtp_timestamp();
// Send more SRs to ensure slope is always exact for different offsets
for (int packets = 1; packets <= kExtraPackets; ++packets) {
clock_.AdvanceTimeMicroseconds(kTimeBetweenSRsUs);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr));
EXPECT_EQ(packets + 1, parser()->sender_report()->num_packets());
NtpTime ntp2 = parser()->sender_report()->ntp();
uint32_t rtp2 = parser()->sender_report()->rtp_timestamp();
uint32_t ntp_diff_in_rtp_units =
(ntp2.ToMs() - ntp1.ToMs()) * (kVideoPayloadTypeFrequency / 1000);
EXPECT_EQ(rtp2 - rtp1, ntp_diff_in_rtp_units);
}
}
TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
nullptr, nullptr, &test_transport_, 1000));
rtcp_sender_->SetSSRC(kSenderSsrc);
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender_->SetSendingStatus(feedback_state(), true);
// Sender Report shouldn't be send as an SR nor as a Report.
rtcp_sender_->SendRTCP(feedback_state(), kRtcpSr);
EXPECT_EQ(0, parser()->sender_report()->num_packets());
rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport);
EXPECT_EQ(0, parser()->sender_report()->num_packets());
// Other packets (e.g. Pli) are allowed, even if useless.
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
EXPECT_EQ(1, parser()->pli()->num_packets());
}
TEST_F(RtcpSenderTest, DoNotSendCompundBeforeRtp) {
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
nullptr, nullptr, &test_transport_, 1000));
rtcp_sender_->SetSSRC(kSenderSsrc);
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
rtcp_sender_->SetSendingStatus(feedback_state(), true);
// In compound mode no packets are allowed (e.g. Pli) because compound mode
// should start with Sender Report.
EXPECT_EQ(-1, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
EXPECT_EQ(0, parser()->pli()->num_packets());
}
TEST_F(RtcpSenderTest, SendRr) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpRr));
EXPECT_EQ(1, parser()->receiver_report()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->receiver_report()->sender_ssrc());
EXPECT_EQ(0U, parser()->receiver_report()->report_blocks().size());
}
TEST_F(RtcpSenderTest, SendRrWithOneReportBlock) {
const uint16_t kSeqNum = 11111;
InsertIncomingPacket(kRemoteSsrc, kSeqNum);
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpRr));
EXPECT_EQ(1, parser()->receiver_report()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->receiver_report()->sender_ssrc());
ASSERT_EQ(1U, parser()->receiver_report()->report_blocks().size());
const rtcp::ReportBlock& rb = parser()->receiver_report()->report_blocks()[0];
EXPECT_EQ(kRemoteSsrc, rb.source_ssrc());
EXPECT_EQ(0U, rb.fraction_lost());
EXPECT_EQ(0, rb.cumulative_lost_signed());
EXPECT_EQ(kSeqNum, rb.extended_high_seq_num());
}
TEST_F(RtcpSenderTest, SendRrWithTwoReportBlocks) {
const uint16_t kSeqNum = 11111;
InsertIncomingPacket(kRemoteSsrc, kSeqNum);
InsertIncomingPacket(kRemoteSsrc + 1, kSeqNum + 1);
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpRr));
EXPECT_EQ(1, parser()->receiver_report()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->receiver_report()->sender_ssrc());
EXPECT_EQ(2U, parser()->receiver_report()->report_blocks().size());
EXPECT_EQ(kRemoteSsrc,
parser()->receiver_report()->report_blocks()[0].source_ssrc());
EXPECT_EQ(kRemoteSsrc + 1,
parser()->receiver_report()->report_blocks()[1].source_ssrc());
}
TEST_F(RtcpSenderTest, SendSdes) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SetCNAME("alice@host"));
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSdes));
EXPECT_EQ(1, parser()->sdes()->num_packets());
EXPECT_EQ(1U, parser()->sdes()->chunks().size());
EXPECT_EQ(kSenderSsrc, parser()->sdes()->chunks()[0].ssrc);
EXPECT_EQ("alice@host", parser()->sdes()->chunks()[0].cname);
}
TEST_F(RtcpSenderTest, SendSdesWithMaxChunks) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SetCNAME("alice@host"));
const char cname[] = "smith@host";
for (size_t i = 0; i < 30; ++i) {
const uint32_t csrc = 0x1234 + i;
EXPECT_EQ(0, rtcp_sender_->AddMixedCNAME(csrc, cname));
}
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSdes));
EXPECT_EQ(1, parser()->sdes()->num_packets());
EXPECT_EQ(31U, parser()->sdes()->chunks().size());
}
TEST_F(RtcpSenderTest, SdesIncludedInCompoundPacket) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
EXPECT_EQ(0, rtcp_sender_->SetCNAME("alice@host"));
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
EXPECT_EQ(1, parser()->receiver_report()->num_packets());
EXPECT_EQ(1, parser()->sdes()->num_packets());
EXPECT_EQ(1U, parser()->sdes()->chunks().size());
}
TEST_F(RtcpSenderTest, SendBye) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye));
EXPECT_EQ(1, parser()->bye()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->bye()->sender_ssrc());
}
TEST_F(RtcpSenderTest, StopSendingTriggersBye) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), true));
EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
EXPECT_EQ(1, parser()->bye()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->bye()->sender_ssrc());
}
TEST_F(RtcpSenderTest, SendApp) {
const uint8_t kSubType = 30;
uint32_t name = 'n' << 24;
name += 'a' << 16;
name += 'm' << 8;
name += 'e';
const uint8_t kData[] = {'t', 'e', 's', 't', 'd', 'a', 't', 'a'};
EXPECT_EQ(0, rtcp_sender_->SetApplicationSpecificData(kSubType, name, kData,
sizeof(kData)));
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpApp));
EXPECT_EQ(1, parser()->app()->num_packets());
EXPECT_EQ(kSubType, parser()->app()->sub_type());
EXPECT_EQ(name, parser()->app()->name());
EXPECT_EQ(sizeof(kData), parser()->app()->data_size());
EXPECT_EQ(0, memcmp(kData, parser()->app()->data(), sizeof(kData)));
}
TEST_F(RtcpSenderTest, SendEmptyApp) {
const uint8_t kSubType = 30;
const uint32_t kName = 0x6E616D65;
EXPECT_EQ(
0, rtcp_sender_->SetApplicationSpecificData(kSubType, kName, nullptr, 0));
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpApp));
EXPECT_EQ(1, parser()->app()->num_packets());
EXPECT_EQ(kSubType, parser()->app()->sub_type());
EXPECT_EQ(kName, parser()->app()->name());
EXPECT_EQ(0U, parser()->app()->data_size());
}
TEST_F(RtcpSenderTest, SetInvalidApplicationSpecificData) {
const uint8_t kData[] = {'t', 'e', 's', 't', 'd', 'a', 't'};
const uint16_t kInvalidDataLength = sizeof(kData) / sizeof(kData[0]);
EXPECT_EQ(-1,
rtcp_sender_->SetApplicationSpecificData(
0, 0, kData, kInvalidDataLength)); // Should by multiple of 4.
}
TEST_F(RtcpSenderTest, SendFir) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpFir));
EXPECT_EQ(1, parser()->fir()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->fir()->sender_ssrc());
EXPECT_EQ(1U, parser()->fir()->requests().size());
EXPECT_EQ(kRemoteSsrc, parser()->fir()->requests()[0].ssrc);
uint8_t seq = parser()->fir()->requests()[0].seq_nr;
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpFir));
EXPECT_EQ(2, parser()->fir()->num_packets());
EXPECT_EQ(seq + 1, parser()->fir()->requests()[0].seq_nr);
}
TEST_F(RtcpSenderTest, SendPli) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
EXPECT_EQ(1, parser()->pli()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->pli()->sender_ssrc());
EXPECT_EQ(kRemoteSsrc, parser()->pli()->media_ssrc());
}
TEST_F(RtcpSenderTest, SendNack) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
const uint16_t kList[] = {0, 1, 16};
const int32_t kListLength = sizeof(kList) / sizeof(kList[0]);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpNack, kListLength,
kList));
EXPECT_EQ(1, parser()->nack()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->nack()->sender_ssrc());
EXPECT_EQ(kRemoteSsrc, parser()->nack()->media_ssrc());
EXPECT_THAT(parser()->nack()->packet_ids(), ElementsAre(0, 1, 16));
}
TEST_F(RtcpSenderTest, RembNotIncludedBeforeSet) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender_->SendRTCP(feedback_state(), kRtcpRr);
ASSERT_EQ(1, parser()->receiver_report()->num_packets());
EXPECT_EQ(0, parser()->remb()->num_packets());
}
TEST_F(RtcpSenderTest, RembNotIncludedAfterUnset) {
const uint64_t kBitrate = 261011;
const std::vector<uint32_t> kSsrcs = {kRemoteSsrc, kRemoteSsrc + 1};
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender_->SetRemb(kBitrate, kSsrcs);
rtcp_sender_->SendRTCP(feedback_state(), kRtcpRr);
ASSERT_EQ(1, parser()->receiver_report()->num_packets());
EXPECT_EQ(1, parser()->remb()->num_packets());
// Turn off REMB. rtcp_sender no longer should send it.
rtcp_sender_->UnsetRemb();
rtcp_sender_->SendRTCP(feedback_state(), kRtcpRr);
ASSERT_EQ(2, parser()->receiver_report()->num_packets());
EXPECT_EQ(1, parser()->remb()->num_packets());
}
TEST_F(RtcpSenderTest, SendRemb) {
const uint64_t kBitrate = 261011;
const std::vector<uint32_t> kSsrcs = {kRemoteSsrc, kRemoteSsrc + 1};
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender_->SetRemb(kBitrate, kSsrcs);
rtcp_sender_->SendRTCP(feedback_state(), kRtcpRemb);
EXPECT_EQ(1, parser()->remb()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->remb()->sender_ssrc());
EXPECT_EQ(kBitrate, parser()->remb()->bitrate_bps());
EXPECT_THAT(parser()->remb()->ssrcs(),
ElementsAre(kRemoteSsrc, kRemoteSsrc + 1));
}
TEST_F(RtcpSenderTest, RembIncludedInEachCompoundPacketAfterSet) {
const int kBitrate = 261011;
const std::vector<uint32_t> kSsrcs = {kRemoteSsrc, kRemoteSsrc + 1};
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
rtcp_sender_->SetRemb(kBitrate, kSsrcs);
rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport);
EXPECT_EQ(1, parser()->remb()->num_packets());
// REMB should be included in each compound packet.
rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport);
EXPECT_EQ(2, parser()->remb()->num_packets());
}
TEST_F(RtcpSenderTest, SendXrWithDlrr) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
rtcp::ReceiveTimeInfo last_xr_rr;
last_xr_rr.ssrc = 0x11111111;
last_xr_rr.last_rr = 0x22222222;
last_xr_rr.delay_since_last_rr = 0x33333333;
feedback_state.last_xr_rtis.push_back(last_xr_rr);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport));
EXPECT_EQ(1, parser()->xr()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
ASSERT_THAT(parser()->xr()->dlrr().sub_blocks(), SizeIs(1));
EXPECT_EQ(last_xr_rr.ssrc, parser()->xr()->dlrr().sub_blocks()[0].ssrc);
EXPECT_EQ(last_xr_rr.last_rr, parser()->xr()->dlrr().sub_blocks()[0].last_rr);
EXPECT_EQ(last_xr_rr.delay_since_last_rr,
parser()->xr()->dlrr().sub_blocks()[0].delay_since_last_rr);
}
TEST_F(RtcpSenderTest, SendXrWithMultipleDlrrSubBlocks) {
const size_t kNumReceivers = 2;
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
for (size_t i = 0; i < kNumReceivers; ++i) {
rtcp::ReceiveTimeInfo last_xr_rr;
last_xr_rr.ssrc = i;
last_xr_rr.last_rr = (i + 1) * 100;
last_xr_rr.delay_since_last_rr = (i + 2) * 200;
feedback_state.last_xr_rtis.push_back(last_xr_rr);
}
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport));
EXPECT_EQ(1, parser()->xr()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
ASSERT_THAT(parser()->xr()->dlrr().sub_blocks(), SizeIs(kNumReceivers));
for (size_t i = 0; i < kNumReceivers; ++i) {
EXPECT_EQ(feedback_state.last_xr_rtis[i].ssrc,
parser()->xr()->dlrr().sub_blocks()[i].ssrc);
EXPECT_EQ(feedback_state.last_xr_rtis[i].last_rr,
parser()->xr()->dlrr().sub_blocks()[i].last_rr);
EXPECT_EQ(feedback_state.last_xr_rtis[i].delay_since_last_rr,
parser()->xr()->dlrr().sub_blocks()[i].delay_since_last_rr);
}
}
TEST_F(RtcpSenderTest, SendXrWithRrtr) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
rtcp_sender_->SendRtcpXrReceiverReferenceTime(true);
NtpTime ntp = TimeMicrosToNtp(clock_.TimeInMicroseconds());
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
EXPECT_EQ(1, parser()->xr()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
EXPECT_FALSE(parser()->xr()->dlrr());
ASSERT_TRUE(parser()->xr()->rrtr());
EXPECT_EQ(ntp, parser()->xr()->rrtr()->ntp());
}
TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfSending) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), true));
rtcp_sender_->SendRtcpXrReceiverReferenceTime(true);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
EXPECT_EQ(0, parser()->xr()->num_packets());
}
TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfNotEnabled) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
rtcp_sender_->SendRtcpXrReceiverReferenceTime(false);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
EXPECT_EQ(0, parser()->xr()->num_packets());
}
TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
RtcpPacketTypeCounterObserverImpl observer;
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
&observer, nullptr, &test_transport_,
1000));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
EXPECT_EQ(1, parser()->pli()->num_packets());
EXPECT_EQ(kRemoteSsrc, observer.ssrc_);
EXPECT_EQ(1U, observer.counter_.pli_packets);
EXPECT_EQ(clock_.TimeInMilliseconds(),
observer.counter_.first_packet_time_ms);
}
TEST_F(RtcpSenderTest, SendTmmbr) {
const unsigned int kBitrateBps = 312000;
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender_->SetTargetBitrate(kBitrateBps);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpTmmbr));
EXPECT_EQ(1, parser()->tmmbr()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->tmmbr()->sender_ssrc());
EXPECT_EQ(1U, parser()->tmmbr()->requests().size());
EXPECT_EQ(kBitrateBps, parser()->tmmbr()->requests()[0].bitrate_bps());
// TODO(asapersson): tmmbr_item()->Overhead() looks broken, always zero.
}
TEST_F(RtcpSenderTest, TmmbrIncludedInCompoundPacketIfEnabled) {
const unsigned int kBitrateBps = 312000;
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
EXPECT_FALSE(rtcp_sender_->TMMBR());
rtcp_sender_->SetTMMBRStatus(true);
EXPECT_TRUE(rtcp_sender_->TMMBR());
rtcp_sender_->SetTargetBitrate(kBitrateBps);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
EXPECT_EQ(1, parser()->tmmbr()->num_packets());
EXPECT_EQ(1U, parser()->tmmbr()->requests().size());
// TMMBR should be included in each compound packet.
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
EXPECT_EQ(2, parser()->tmmbr()->num_packets());
rtcp_sender_->SetTMMBRStatus(false);
EXPECT_FALSE(rtcp_sender_->TMMBR());
}
TEST_F(RtcpSenderTest, SendTmmbn) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
rtcp_sender_->SetSendingStatus(feedback_state(), true);
std::vector<rtcp::TmmbItem> bounding_set;
const uint32_t kBitrateBps = 32768000;
const uint32_t kPacketOh = 40;
const uint32_t kSourceSsrc = 12345;
const rtcp::TmmbItem tmmbn(kSourceSsrc, kBitrateBps, kPacketOh);
bounding_set.push_back(tmmbn);
rtcp_sender_->SetTmmbn(bounding_set);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSr));
EXPECT_EQ(1, parser()->sender_report()->num_packets());
EXPECT_EQ(1, parser()->tmmbn()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->tmmbn()->sender_ssrc());
EXPECT_EQ(1U, parser()->tmmbn()->items().size());
EXPECT_EQ(kBitrateBps, parser()->tmmbn()->items()[0].bitrate_bps());
EXPECT_EQ(kPacketOh, parser()->tmmbn()->items()[0].packet_overhead());
EXPECT_EQ(kSourceSsrc, parser()->tmmbn()->items()[0].ssrc());
}
// This test is written to verify actual behaviour. It does not seem
// to make much sense to send an empty TMMBN, since there is no place
// to put an actual limit here. It's just information that no limit
// is set, which is kind of the starting assumption.
// See http://code.google.com/p/webrtc/issues/detail?id=468 for one
// situation where this caused confusion.
TEST_F(RtcpSenderTest, SendsTmmbnIfSetAndEmpty) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
rtcp_sender_->SetSendingStatus(feedback_state(), true);
std::vector<rtcp::TmmbItem> bounding_set;
rtcp_sender_->SetTmmbn(bounding_set);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSr));
EXPECT_EQ(1, parser()->sender_report()->num_packets());
EXPECT_EQ(1, parser()->tmmbn()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->tmmbn()->sender_ssrc());
EXPECT_EQ(0U, parser()->tmmbn()->items().size());
}
TEST_F(RtcpSenderTest, SendCompoundPliRemb) {
const int kBitrate = 261011;
std::vector<uint32_t> ssrcs;
ssrcs.push_back(kRemoteSsrc);
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
rtcp_sender_->SetRemb(kBitrate, ssrcs);
std::set<RTCPPacketType> packet_types;
packet_types.insert(kRtcpRemb);
packet_types.insert(kRtcpPli);
EXPECT_EQ(0, rtcp_sender_->SendCompoundRTCP(feedback_state(), packet_types));
EXPECT_EQ(1, parser()->remb()->num_packets());
EXPECT_EQ(1, parser()->pli()->num_packets());
}
// This test is written to verify that BYE is always the last packet
// type in a RTCP compoud packet. The rtcp_sender_ is recreated with
// mock_transport, which is used to check for whether BYE at the end
// of a RTCP compound packet.
TEST_F(RtcpSenderTest, ByeMustBeLast) {
MockTransport mock_transport;
EXPECT_CALL(mock_transport, SendRtcp(_, _))
.WillOnce(Invoke([](const uint8_t* data, size_t len) {
const uint8_t* next_packet = data;
const uint8_t* const packet_end = data + len;
rtcp::CommonHeader packet;
while (next_packet < packet_end) {
EXPECT_TRUE(packet.Parse(next_packet, packet_end - next_packet));
next_packet = packet.NextPacket();
if (packet.type() ==
rtcp::Bye::kPacketType) // Main test expectation.
EXPECT_EQ(0, packet_end - next_packet)
<< "Bye packet should be last in a compound RTCP packet.";
if (next_packet == packet_end) // Validate test was set correctly.
EXPECT_EQ(packet.type(), rtcp::Bye::kPacketType)
<< "Last packet in this test expected to be Bye.";
}
return true;
}));
// Re-configure rtcp_sender_ with mock_transport_
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
nullptr, nullptr, &mock_transport, 1000));
rtcp_sender_->SetSSRC(kSenderSsrc);
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds(),
/*paylpad_type=*/0);
// Set up REMB info to be included with BYE.
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
rtcp_sender_->SetRemb(1234, {});
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye));
}
TEST_F(RtcpSenderTest, SendXrWithTargetBitrate) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
const size_t kNumSpatialLayers = 2;
const size_t kNumTemporalLayers = 2;
VideoBitrateAllocation allocation;
for (size_t sl = 0; sl < kNumSpatialLayers; ++sl) {
uint32_t start_bitrate_bps = (sl + 1) * 100000;
for (size_t tl = 0; tl < kNumTemporalLayers; ++tl)
allocation.SetBitrate(sl, tl, start_bitrate_bps + (tl * 20000));
}
rtcp_sender_->SetVideoBitrateAllocation(allocation);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
EXPECT_EQ(1, parser()->xr()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
const absl::optional<rtcp::TargetBitrate>& target_bitrate =
parser()->xr()->target_bitrate();
ASSERT_TRUE(target_bitrate);
const std::vector<rtcp::TargetBitrate::BitrateItem>& bitrates =
target_bitrate->GetTargetBitrates();
EXPECT_EQ(kNumSpatialLayers * kNumTemporalLayers, bitrates.size());
for (size_t sl = 0; sl < kNumSpatialLayers; ++sl) {
uint32_t start_bitrate_bps = (sl + 1) * 100000;
for (size_t tl = 0; tl < kNumTemporalLayers; ++tl) {
size_t index = (sl * kNumSpatialLayers) + tl;
const rtcp::TargetBitrate::BitrateItem& item = bitrates[index];
EXPECT_EQ(sl, item.spatial_layer);
EXPECT_EQ(tl, item.temporal_layer);
EXPECT_EQ(start_bitrate_bps + (tl * 20000),
item.target_bitrate_kbps * 1000);
}
}
}
TEST_F(RtcpSenderTest, SendImmediateXrWithTargetBitrate) {
// Initialize. Send a first report right away.
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
clock_.AdvanceTimeMilliseconds(5);
// Video bitrate allocation generated, save until next time we send a report.
VideoBitrateAllocation allocation;
allocation.SetBitrate(0, 0, 100000);
rtcp_sender_->SetVideoBitrateAllocation(allocation);
// First seen instance will be sent immediately.
EXPECT_TRUE(rtcp_sender_->TimeToSendRTCPReport(false));
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
clock_.AdvanceTimeMilliseconds(5);
// Update bitrate of existing layer, does not quality for immediate sending.
allocation.SetBitrate(0, 0, 150000);
rtcp_sender_->SetVideoBitrateAllocation(allocation);
EXPECT_FALSE(rtcp_sender_->TimeToSendRTCPReport(false));
// A new spatial layer enabled, signal this as soon as possible.
allocation.SetBitrate(1, 0, 200000);
rtcp_sender_->SetVideoBitrateAllocation(allocation);
EXPECT_TRUE(rtcp_sender_->TimeToSendRTCPReport(false));
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
clock_.AdvanceTimeMilliseconds(5);
// Explicitly disable top layer. The same set of layers now has a bitrate
// defined, but the explicit 0 indicates shutdown. Signal immediately.
allocation.SetBitrate(1, 0, 0);
EXPECT_FALSE(rtcp_sender_->TimeToSendRTCPReport(false));
rtcp_sender_->SetVideoBitrateAllocation(allocation);
EXPECT_TRUE(rtcp_sender_->TimeToSendRTCPReport(false));
}
TEST_F(RtcpSenderTest, SendTargetBitrateExplicitZeroOnStreamRemoval) {
// Set up and send a bitrate allocation with two layers.
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
VideoBitrateAllocation allocation;
allocation.SetBitrate(0, 0, 100000);
allocation.SetBitrate(1, 0, 200000);
rtcp_sender_->SetVideoBitrateAllocation(allocation);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
absl::optional<rtcp::TargetBitrate> target_bitrate =
parser()->xr()->target_bitrate();
ASSERT_TRUE(target_bitrate);
std::vector<rtcp::TargetBitrate::BitrateItem> bitrates =
target_bitrate->GetTargetBitrates();
ASSERT_EQ(2u, bitrates.size());
EXPECT_EQ(bitrates[0].target_bitrate_kbps,
allocation.GetBitrate(0, 0) / 1000);
EXPECT_EQ(bitrates[1].target_bitrate_kbps,
allocation.GetBitrate(1, 0) / 1000);
// Create a new allocation, where the second stream is no longer available.
VideoBitrateAllocation new_allocation;
new_allocation.SetBitrate(0, 0, 150000);
rtcp_sender_->SetVideoBitrateAllocation(new_allocation);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
target_bitrate = parser()->xr()->target_bitrate();
ASSERT_TRUE(target_bitrate);
bitrates = target_bitrate->GetTargetBitrates();
// Two bitrates should still be set, with an explicit entry indicating the
// removed stream is gone.
ASSERT_EQ(2u, bitrates.size());
EXPECT_EQ(bitrates[0].target_bitrate_kbps,
new_allocation.GetBitrate(0, 0) / 1000);
EXPECT_EQ(bitrates[1].target_bitrate_kbps, 0u);
}
} // namespace webrtc