blob: 65e2e04ef44da23b6d5d2f74d259ba20d56086f6 [file] [log] [blame]
/*
* 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 "modules/rtp_rtcp/source/rtp_sender.h"
#include <memory>
#include <vector>
#include "api/rtc_event_log/rtc_event.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_codec_constants.h"
#include "api/video/video_timing.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/rtp_rtcp/source/rtp_sender_egress.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/strings/string_builder.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_transport.h"
#include "test/rtp_header_parser.h"
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kAbsoluteSendTimeExtensionId = 1,
kAudioLevelExtensionId,
kGenericDescriptorId,
kMidExtensionId,
kRepairedRidExtensionId,
kRidExtensionId,
kTransmissionTimeOffsetExtensionId,
kTransportSequenceNumberExtensionId,
kVideoRotationExtensionId,
kVideoTimingExtensionId,
};
const int kPayload = 100;
const int kRtxPayload = 98;
const uint32_t kTimestamp = 10;
const uint16_t kSeqNum = 33;
const uint32_t kSsrc = 725242;
const uint32_t kRtxSsrc = 12345;
const uint32_t kFlexFecSsrc = 45678;
const uint16_t kTransportSequenceNumber = 1;
const uint64_t kStartTime = 123456789;
const size_t kMaxPaddingSize = 224u;
const uint8_t kPayloadData[] = {47, 11, 32, 93, 89};
const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
const char kNoRid[] = "";
const char kNoMid[] = "";
using ::testing::_;
using ::testing::AllOf;
using ::testing::Contains;
using ::testing::Each;
using ::testing::ElementsAreArray;
using ::testing::Eq;
using ::testing::Field;
using ::testing::Gt;
using ::testing::IsEmpty;
using ::testing::NiceMock;
using ::testing::Not;
using ::testing::Pointee;
using ::testing::Property;
using ::testing::Return;
using ::testing::SizeIs;
using ::testing::StrictMock;
uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;
}
class LoopbackTransportTest : public webrtc::Transport {
public:
LoopbackTransportTest() : total_bytes_sent_(0) {
receivers_extensions_.Register<TransmissionOffset>(
kTransmissionTimeOffsetExtensionId);
receivers_extensions_.Register<AbsoluteSendTime>(
kAbsoluteSendTimeExtensionId);
receivers_extensions_.Register<TransportSequenceNumber>(
kTransportSequenceNumberExtensionId);
receivers_extensions_.Register<VideoOrientation>(kVideoRotationExtensionId);
receivers_extensions_.Register<AudioLevel>(kAudioLevelExtensionId);
receivers_extensions_.Register<VideoTimingExtension>(
kVideoTimingExtensionId);
receivers_extensions_.Register<RtpMid>(kMidExtensionId);
receivers_extensions_.Register<RtpGenericFrameDescriptorExtension00>(
kGenericDescriptorId);
receivers_extensions_.Register<RtpStreamId>(kRidExtensionId);
receivers_extensions_.Register<RepairedRtpStreamId>(
kRepairedRidExtensionId);
}
bool SendRtp(const uint8_t* data,
size_t len,
const PacketOptions& options) override {
last_options_ = options;
total_bytes_sent_ += len;
sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
EXPECT_TRUE(sent_packets_.back().Parse(data, len));
return true;
}
bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); }
int packets_sent() { return sent_packets_.size(); }
size_t total_bytes_sent_;
PacketOptions last_options_;
std::vector<RtpPacketReceived> sent_packets_;
private:
RtpHeaderExtensionMap receivers_extensions_;
};
MATCHER_P(SameRtcEventTypeAs, value, "") {
return value == arg->GetType();
}
struct TestConfig {
explicit TestConfig(bool with_overhead) : with_overhead(with_overhead) {}
bool with_overhead = false;
};
class MockRtpPacketPacer : public RtpPacketSender {
public:
MockRtpPacketPacer() {}
virtual ~MockRtpPacketPacer() {}
MOCK_METHOD1(EnqueuePackets,
void(std::vector<std::unique_ptr<RtpPacketToSend>>));
MOCK_METHOD2(CreateProbeCluster, void(int bitrate_bps, int cluster_id));
MOCK_METHOD0(Pause, void());
MOCK_METHOD0(Resume, void());
MOCK_METHOD1(SetCongestionWindow,
void(absl::optional<int64_t> congestion_window_bytes));
MOCK_METHOD1(UpdateOutstandingData, void(int64_t outstanding_bytes));
MOCK_METHOD1(SetAccountForAudioPackets, void(bool account_for_audio));
};
class MockSendSideDelayObserver : public SendSideDelayObserver {
public:
MOCK_METHOD4(SendSideDelayUpdated, void(int, int, uint64_t, uint32_t));
};
class MockSendPacketObserver : public SendPacketObserver {
public:
MOCK_METHOD3(OnSendPacket, void(uint16_t, int64_t, uint32_t));
};
class MockTransportFeedbackObserver : public TransportFeedbackObserver {
public:
MOCK_METHOD1(OnAddPacket, void(const RtpPacketSendInfo&));
MOCK_METHOD1(OnTransportFeedback, void(const rtcp::TransportFeedback&));
};
class StreamDataTestCallback : public StreamDataCountersCallback {
public:
StreamDataTestCallback()
: StreamDataCountersCallback(), ssrc_(0), counters_() {}
~StreamDataTestCallback() override = default;
void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) override {
ssrc_ = ssrc;
counters_ = counters;
}
uint32_t ssrc_;
StreamDataCounters counters_;
void MatchPacketCounter(const RtpPacketCounter& expected,
const RtpPacketCounter& actual) {
EXPECT_EQ(expected.payload_bytes, actual.payload_bytes);
EXPECT_EQ(expected.header_bytes, actual.header_bytes);
EXPECT_EQ(expected.padding_bytes, actual.padding_bytes);
EXPECT_EQ(expected.packets, actual.packets);
}
void Matches(uint32_t ssrc, const StreamDataCounters& counters) {
EXPECT_EQ(ssrc, ssrc_);
MatchPacketCounter(counters.transmitted, counters_.transmitted);
MatchPacketCounter(counters.retransmitted, counters_.retransmitted);
EXPECT_EQ(counters.fec.packets, counters_.fec.packets);
}
};
// Mimics ModuleRtpRtcp::RtpSenderContext.
// TODO(sprang): Split up unit tests and test these components individually
// wherever possible.
struct RtpSenderContext {
explicit RtpSenderContext(const RtpRtcp::Configuration& config)
: packet_history_(config.clock, config.enable_rtx_padding_prioritization),
packet_sender_(config, &packet_history_),
non_paced_sender_(&packet_sender_),
packet_generator_(
config,
&packet_history_,
config.paced_sender ? config.paced_sender : &non_paced_sender_) {}
RtpPacketHistory packet_history_;
RtpSenderEgress packet_sender_;
RtpSenderEgress::NonPacedPacketSender non_paced_sender_;
RTPSender packet_generator_;
};
class FieldTrialConfig : public WebRtcKeyValueConfig {
public:
FieldTrialConfig() : overhead_enabled_(false), max_padding_factor_(1200) {}
~FieldTrialConfig() override {}
void SetOverHeadEnabled(bool enabled) { overhead_enabled_ = enabled; }
void SetMaxPaddingFactor(double factor) { max_padding_factor_ = factor; }
std::string Lookup(absl::string_view key) const override {
if (key == "WebRTC-LimitPaddingSize") {
char string_buf[32];
rtc::SimpleStringBuilder ssb(string_buf);
ssb << "factor:" << max_padding_factor_;
return ssb.str();
} else if (key == "WebRTC-SendSideBwe-WithOverhead") {
return overhead_enabled_ ? "Enabled" : "Disabled";
}
return "";
}
private:
bool overhead_enabled_;
double max_padding_factor_;
};
} // namespace
class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
protected:
RtpSenderTest()
: fake_clock_(kStartTime),
retransmission_rate_limiter_(&fake_clock_, 1000),
flexfec_sender_(0,
kFlexFecSsrc,
kSsrc,
"",
std::vector<RtpExtension>(),
std::vector<RtpExtensionSize>(),
nullptr,
&fake_clock_),
kMarkerBit(true) {
field_trials_.SetOverHeadEnabled(GetParam().with_overhead);
}
void SetUp() override { SetUpRtpSender(true, false, false); }
RTPSender* rtp_sender() {
RTC_DCHECK(rtp_sender_context_);
return &rtp_sender_context_->packet_generator_;
}
RtpSenderEgress* rtp_egress() {
RTC_DCHECK(rtp_sender_context_);
return &rtp_sender_context_->packet_sender_;
}
void SetUpRtpSender(bool pacer,
bool populate_network2,
bool always_send_mid_and_rid) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.rtx_send_ssrc = kRtxSsrc;
config.fec_generator = &flexfec_sender_;
config.event_log = &mock_rtc_event_log_;
config.send_packet_observer = &send_packet_observer_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.paced_sender = pacer ? &mock_paced_sender_ : nullptr;
config.populate_network2_timestamp = populate_network2;
config.rtp_stats_callback = &rtp_stats_callback_;
config.always_send_mid_and_rid = always_send_mid_and_rid;
config.field_trials = &field_trials_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
rtp_sender()->SetSequenceNumber(kSeqNum);
rtp_sender()->SetTimestampOffset(0);
}
SimulatedClock fake_clock_;
NiceMock<MockRtcEventLog> mock_rtc_event_log_;
MockRtpPacketPacer mock_paced_sender_;
StrictMock<MockSendPacketObserver> send_packet_observer_;
StrictMock<MockTransportFeedbackObserver> feedback_observer_;
RateLimiter retransmission_rate_limiter_;
FlexfecSender flexfec_sender_;
std::unique_ptr<RtpSenderContext> rtp_sender_context_;
LoopbackTransportTest transport_;
const bool kMarkerBit;
FieldTrialConfig field_trials_;
StreamDataTestCallback rtp_stats_callback_;
std::unique_ptr<RtpPacketToSend> BuildRtpPacket(int payload_type,
bool marker_bit,
uint32_t timestamp,
int64_t capture_time_ms) {
auto packet = rtp_sender()->AllocatePacket();
packet->SetPayloadType(payload_type);
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->SetMarker(marker_bit);
packet->SetTimestamp(timestamp);
packet->set_capture_time_ms(capture_time_ms);
EXPECT_TRUE(rtp_sender()->AssignSequenceNumber(packet.get()));
return packet;
}
std::unique_ptr<RtpPacketToSend> SendPacket(int64_t capture_time_ms,
int payload_length) {
uint32_t timestamp = capture_time_ms * 90;
auto packet =
BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
packet->AllocatePayload(payload_length);
packet->set_allow_retransmission(true);
// Packet should be stored in a send bucket.
EXPECT_TRUE(rtp_sender()->SendToNetwork(
std::make_unique<RtpPacketToSend>(*packet)));
return packet;
}
std::unique_ptr<RtpPacketToSend> SendGenericPacket() {
const int64_t kCaptureTimeMs = fake_clock_.TimeInMilliseconds();
return SendPacket(kCaptureTimeMs, sizeof(kPayloadData));
}
size_t GenerateAndSendPadding(size_t target_size_bytes) {
size_t generated_bytes = 0;
for (auto& packet :
rtp_sender()->GeneratePadding(target_size_bytes, true)) {
generated_bytes += packet->payload_size() + packet->padding_size();
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
}
return generated_bytes;
}
// The following are helpers for configuring the RTPSender. They must be
// called before sending any packets.
// Enable the retransmission stream with sizable packet storage.
void EnableRtx() {
// RTX needs to be able to read the source packets from the packet store.
// Pick a number of packets to store big enough for any unit test.
constexpr uint16_t kNumberOfPacketsToStore = 100;
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, kNumberOfPacketsToStore);
rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
}
// Enable sending of the MID header extension for both the primary SSRC and
// the RTX SSRC.
void EnableMidSending(const std::string& mid) {
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionMid, kMidExtensionId);
rtp_sender()->SetMid(mid);
}
// Enable sending of the RSID header extension for the primary SSRC and the
// RRSID header extension for the RTX SSRC.
void EnableRidSending(const std::string& rid) {
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionRtpStreamId,
kRidExtensionId);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionRepairedRtpStreamId,
kRepairedRidExtensionId);
rtp_sender()->SetRid(rid);
}
};
// TODO(pbos): Move tests over from WithoutPacer to RtpSenderTest as this is our
// default code path.
class RtpSenderTestWithoutPacer : public RtpSenderTest {
public:
void SetUp() override { SetUpRtpSender(false, false, false); }
};
TEST_P(RtpSenderTestWithoutPacer, AllocatePacketSetCsrc) {
// Configure rtp_sender with csrc.
std::vector<uint32_t> csrcs;
csrcs.push_back(0x23456789);
rtp_sender()->SetCsrcs(csrcs);
auto packet = rtp_sender()->AllocatePacket();
ASSERT_TRUE(packet);
EXPECT_EQ(rtp_sender()->SSRC(), packet->Ssrc());
EXPECT_EQ(csrcs, packet->Csrcs());
}
TEST_P(RtpSenderTestWithoutPacer, AllocatePacketReserveExtensions) {
// Configure rtp_sender with extensions.
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
ASSERT_EQ(0,
rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, kAudioLevelExtensionId));
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionVideoRotation, kVideoRotationExtensionId));
auto packet = rtp_sender()->AllocatePacket();
ASSERT_TRUE(packet);
// Preallocate BWE extensions RtpSender set itself.
EXPECT_TRUE(packet->HasExtension<TransmissionOffset>());
EXPECT_TRUE(packet->HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(packet->HasExtension<TransportSequenceNumber>());
// Do not allocate media specific extensions.
EXPECT_FALSE(packet->HasExtension<AudioLevel>());
EXPECT_FALSE(packet->HasExtension<VideoOrientation>());
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberAdvanceSequenceNumber) {
auto packet = rtp_sender()->AllocatePacket();
ASSERT_TRUE(packet);
const uint16_t sequence_number = rtp_sender()->SequenceNumber();
EXPECT_TRUE(rtp_sender()->AssignSequenceNumber(packet.get()));
EXPECT_EQ(sequence_number, packet->SequenceNumber());
EXPECT_EQ(sequence_number + 1, rtp_sender()->SequenceNumber());
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberFailsOnNotSending) {
auto packet = rtp_sender()->AllocatePacket();
ASSERT_TRUE(packet);
rtp_sender()->SetSendingMediaStatus(false);
EXPECT_FALSE(rtp_sender()->AssignSequenceNumber(packet.get()));
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPaddingOnVideo) {
constexpr size_t kPaddingSize = 100;
auto packet = rtp_sender()->AllocatePacket();
ASSERT_TRUE(packet);
ASSERT_TRUE(rtp_sender()->GeneratePadding(kPaddingSize, true).empty());
packet->SetMarker(false);
ASSERT_TRUE(rtp_sender()->AssignSequenceNumber(packet.get()));
// Packet without marker bit doesn't allow padding on video stream.
ASSERT_TRUE(rtp_sender()->GeneratePadding(kPaddingSize, true).empty());
packet->SetMarker(true);
ASSERT_TRUE(rtp_sender()->AssignSequenceNumber(packet.get()));
// Packet with marker bit allows send padding.
ASSERT_FALSE(rtp_sender()->GeneratePadding(kPaddingSize, true).empty());
}
TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
MockTransport transport;
RtpRtcp::Configuration config;
config.audio = true;
config.clock = &fake_clock_;
config.outgoing_transport = &transport;
config.paced_sender = &mock_paced_sender_;
config.local_media_ssrc = kSsrc;
config.event_log = &mock_rtc_event_log_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
rtp_sender()->SetTimestampOffset(0);
std::unique_ptr<RtpPacketToSend> audio_packet =
rtp_sender()->AllocatePacket();
// Padding on audio stream allowed regardless of marker in the last packet.
audio_packet->SetMarker(false);
audio_packet->SetPayloadType(kPayload);
rtp_sender()->AssignSequenceNumber(audio_packet.get());
const size_t kPaddingSize = 59;
EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _))
.WillOnce(Return(true));
EXPECT_EQ(kPaddingSize, GenerateAndSendPadding(kPaddingSize));
// Requested padding size is too small, will send a larger one.
const size_t kMinPaddingSize = 50;
EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
.WillOnce(Return(true));
EXPECT_EQ(kMinPaddingSize, GenerateAndSendPadding(kMinPaddingSize - 5));
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
constexpr size_t kPaddingSize = 100;
auto packet = rtp_sender()->AllocatePacket();
ASSERT_TRUE(packet);
packet->SetMarker(true);
packet->SetTimestamp(kTimestamp);
ASSERT_TRUE(rtp_sender()->AssignSequenceNumber(packet.get()));
auto padding_packets = rtp_sender()->GeneratePadding(kPaddingSize, true);
ASSERT_EQ(1u, padding_packets.size());
// Verify padding packet timestamp.
EXPECT_EQ(kTimestamp, padding_packets[0]->Timestamp());
}
TEST_P(RtpSenderTestWithoutPacer,
TransportFeedbackObserverGetsCorrectByteCount) {
constexpr size_t kRtpOverheadBytesPerPacket = 12 + 8;
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.transport_feedback_callback = &feedback_observer_;
config.event_log = &mock_rtc_event_log_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.field_trials = &field_trials_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
const size_t expected_bytes =
GetParam().with_overhead
? sizeof(kPayloadData) + kRtpOverheadBytesPerPacket
: sizeof(kPayloadData);
EXPECT_CALL(feedback_observer_,
OnAddPacket(AllOf(
Field(&RtpPacketSendInfo::ssrc, rtp_sender()->SSRC()),
Field(&RtpPacketSendInfo::transport_sequence_number,
kTransportSequenceNumber),
Field(&RtpPacketSendInfo::rtp_sequence_number,
rtp_sender()->SequenceNumber()),
Field(&RtpPacketSendInfo::length, expected_bytes),
Field(&RtpPacketSendInfo::pacing_info, PacedPacketInfo()))))
.Times(1);
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(),
kRtpOverheadBytesPerPacket);
SendGenericPacket();
}
TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.transport_feedback_callback = &feedback_observer_;
config.event_log = &mock_rtc_event_log_;
config.send_packet_observer = &send_packet_observer_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
EXPECT_CALL(send_packet_observer_,
OnSendPacket(kTransportSequenceNumber, _, _))
.Times(1);
EXPECT_CALL(feedback_observer_,
OnAddPacket(AllOf(
Field(&RtpPacketSendInfo::ssrc, rtp_sender()->SSRC()),
Field(&RtpPacketSendInfo::transport_sequence_number,
kTransportSequenceNumber),
Field(&RtpPacketSendInfo::rtp_sequence_number,
rtp_sender()->SequenceNumber()),
Field(&RtpPacketSendInfo::pacing_info, PacedPacketInfo()))))
.Times(1);
SendGenericPacket();
const auto& packet = transport_.last_sent_packet();
uint16_t transport_seq_no;
ASSERT_TRUE(packet.GetExtension<TransportSequenceNumber>(&transport_seq_no));
EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
EXPECT_EQ(transport_.last_options_.packet_id, transport_seq_no);
EXPECT_TRUE(transport_.last_options_.included_in_allocation);
}
TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.transport_feedback_callback = &feedback_observer_;
config.event_log = &mock_rtc_event_log_;
config.send_packet_observer = &send_packet_observer_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
SendGenericPacket();
EXPECT_FALSE(transport_.last_options_.is_retransmit);
}
TEST_P(RtpSenderTestWithoutPacer,
SetsIncludedInFeedbackWhenTransportSequenceNumberExtensionIsRegistered) {
SetUpRtpSender(false, false, false);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
SendGenericPacket();
EXPECT_TRUE(transport_.last_options_.included_in_feedback);
}
TEST_P(
RtpSenderTestWithoutPacer,
SetsIncludedInAllocationWhenTransportSequenceNumberExtensionIsRegistered) {
SetUpRtpSender(false, false, false);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
SendGenericPacket();
EXPECT_TRUE(transport_.last_options_.included_in_allocation);
}
TEST_P(RtpSenderTestWithoutPacer,
SetsIncludedInAllocationWhenForcedAsPartOfAllocation) {
SetUpRtpSender(false, false, false);
rtp_egress()->ForceIncludeSendPacketsInAllocation(true);
SendGenericPacket();
EXPECT_FALSE(transport_.last_options_.included_in_feedback);
EXPECT_TRUE(transport_.last_options_.included_in_allocation);
}
TEST_P(RtpSenderTestWithoutPacer, DoesnSetIncludedInAllocationByDefault) {
SetUpRtpSender(false, false, false);
SendGenericPacket();
EXPECT_FALSE(transport_.last_options_.included_in_feedback);
EXPECT_FALSE(transport_.last_options_.included_in_allocation);
}
TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
StrictMock<MockSendSideDelayObserver> send_side_delay_observer_;
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.send_side_delay_observer = &send_side_delay_observer_;
config.event_log = &mock_rtc_event_log_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
const uint8_t kPayloadType = 127;
const absl::optional<VideoCodecType> kCodecType =
VideoCodecType::kVideoCodecGeneric;
const uint32_t kCaptureTimeMsToRtpTimestamp = 90; // 90 kHz clock
RTPVideoHeader video_header;
// Send packet with 10 ms send-side delay. The average, max and total should
// be 10 ms.
EXPECT_CALL(send_side_delay_observer_,
SendSideDelayUpdated(10, 10, 10, kSsrc))
.Times(1);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
fake_clock_.AdvanceTimeMilliseconds(10);
video_header.frame_type = VideoFrameType::kVideoFrameKey;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kPayloadType, kCodecType, capture_time_ms * kCaptureTimeMsToRtpTimestamp,
capture_time_ms, kPayloadData, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
// Send another packet with 20 ms delay. The average, max and total should be
// 15, 20 and 30 ms respectively.
EXPECT_CALL(send_side_delay_observer_,
SendSideDelayUpdated(15, 20, 30, kSsrc))
.Times(1);
fake_clock_.AdvanceTimeMilliseconds(10);
video_header.frame_type = VideoFrameType::kVideoFrameKey;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kPayloadType, kCodecType, capture_time_ms * kCaptureTimeMsToRtpTimestamp,
capture_time_ms, kPayloadData, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
// Send another packet at the same time, which replaces the last packet.
// Since this packet has 0 ms delay, the average is now 5 ms and max is 10 ms.
// The total counter stays the same though.
// TODO(terelius): Is is not clear that this is the right behavior.
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(5, 10, 30, kSsrc))
.Times(1);
capture_time_ms = fake_clock_.TimeInMilliseconds();
video_header.frame_type = VideoFrameType::kVideoFrameKey;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kPayloadType, kCodecType, capture_time_ms * kCaptureTimeMsToRtpTimestamp,
capture_time_ms, kPayloadData, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
// Send a packet 1 second later. The earlier packets should have timed
// out, so both max and average should be the delay of this packet. The total
// keeps increasing.
fake_clock_.AdvanceTimeMilliseconds(1000);
capture_time_ms = fake_clock_.TimeInMilliseconds();
fake_clock_.AdvanceTimeMilliseconds(1);
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(1, 1, 31, kSsrc))
.Times(1);
video_header.frame_type = VideoFrameType::kVideoFrameKey;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kPayloadType, kCodecType, capture_time_ms * kCaptureTimeMsToRtpTimestamp,
capture_time_ms, kPayloadData, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
}
TEST_P(RtpSenderTestWithoutPacer, OnSendPacketUpdated) {
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
EXPECT_CALL(send_packet_observer_,
OnSendPacket(kTransportSequenceNumber, _, _))
.Times(1);
SendGenericPacket();
}
TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.paced_sender = &mock_paced_sender_;
config.local_media_ssrc = kSsrc;
config.transport_feedback_callback = &feedback_observer_;
config.event_log = &mock_rtc_event_log_;
config.send_packet_observer = &send_packet_observer_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
rtp_sender()->SetSequenceNumber(kSeqNum);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
EXPECT_CALL(send_packet_observer_,
OnSendPacket(kTransportSequenceNumber, _, _))
.Times(1);
EXPECT_CALL(feedback_observer_,
OnAddPacket(AllOf(
Field(&RtpPacketSendInfo::ssrc, rtp_sender()->SSRC()),
Field(&RtpPacketSendInfo::transport_sequence_number,
kTransportSequenceNumber),
Field(&RtpPacketSendInfo::rtp_sequence_number,
rtp_sender()->SequenceNumber()),
Field(&RtpPacketSendInfo::pacing_info, PacedPacketInfo()))))
.Times(1);
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum))))));
auto packet = SendGenericPacket();
packet->set_packet_type(RtpPacketMediaType::kVideo);
// Transport sequence number is set by PacketRouter, before SendPacket().
packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
uint16_t transport_seq_no;
EXPECT_TRUE(
transport_.last_sent_packet().GetExtension<TransportSequenceNumber>(
&transport_seq_no));
EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
EXPECT_EQ(transport_.last_options_.packet_id, transport_seq_no);
}
TEST_P(RtpSenderTest, WritesPacerExitToTimingExtension) {
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionVideoTiming, kVideoTimingExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet = rtp_sender()->AllocatePacket();
packet->SetPayloadType(kPayload);
packet->SetMarker(true);
packet->SetTimestamp(kTimestamp);
packet->set_capture_time_ms(capture_time_ms);
const VideoSendTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};
packet->SetExtension<VideoTimingExtension>(kVideoTiming);
EXPECT_TRUE(rtp_sender()->AssignSequenceNumber(packet.get()));
size_t packet_size = packet->size();
const int kStoredTimeInMs = 100;
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->set_allow_retransmission(true);
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(Contains(Pointee(Property(
&RtpPacketToSend::Ssrc, kSsrc)))));
EXPECT_TRUE(
rtp_sender()->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
VideoSendTiming video_timing;
EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
&video_timing));
EXPECT_EQ(kStoredTimeInMs, video_timing.pacer_exit_delta_ms);
}
TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtensionWithPacer) {
SetUpRtpSender(/*pacer=*/true, /*populate_network2=*/true, false);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionVideoTiming, kVideoTimingExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet = rtp_sender()->AllocatePacket();
packet->SetPayloadType(kPayload);
packet->SetMarker(true);
packet->SetTimestamp(kTimestamp);
packet->set_capture_time_ms(capture_time_ms);
const uint16_t kPacerExitMs = 1234u;
const VideoSendTiming kVideoTiming = {0u, 0u, 0u, kPacerExitMs, 0u, 0u, true};
packet->SetExtension<VideoTimingExtension>(kVideoTiming);
EXPECT_TRUE(rtp_sender()->AssignSequenceNumber(packet.get()));
size_t packet_size = packet->size();
const int kStoredTimeInMs = 100;
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->set_allow_retransmission(true);
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(Contains(Pointee(Property(
&RtpPacketToSend::Ssrc, kSsrc)))));
EXPECT_TRUE(
rtp_sender()->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
VideoSendTiming video_timing;
EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
&video_timing));
EXPECT_EQ(kStoredTimeInMs, video_timing.network2_timestamp_delta_ms);
EXPECT_EQ(kPacerExitMs, video_timing.pacer_exit_delta_ms);
}
TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtensionWithoutPacer) {
SetUpRtpSender(/*pacer=*/false, /*populate_network2=*/true, false);
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionVideoTiming, kVideoTimingExtensionId));
auto packet = rtp_sender()->AllocatePacket();
packet->SetMarker(true);
packet->set_capture_time_ms(fake_clock_.TimeInMilliseconds());
const VideoSendTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};
packet->SetExtension<VideoTimingExtension>(kVideoTiming);
packet->set_allow_retransmission(true);
EXPECT_TRUE(rtp_sender()->AssignSequenceNumber(packet.get()));
packet->set_packet_type(RtpPacketMediaType::kVideo);
const int kPropagateTimeMs = 10;
fake_clock_.AdvanceTimeMilliseconds(kPropagateTimeMs);
EXPECT_TRUE(rtp_sender()->SendToNetwork(std::move(packet)));
EXPECT_EQ(1, transport_.packets_sent());
absl::optional<VideoSendTiming> video_timing =
transport_.last_sent_packet().GetExtension<VideoTimingExtension>();
ASSERT_TRUE(video_timing);
EXPECT_EQ(kPropagateTimeMs, video_timing->network2_timestamp_delta_ms);
}
TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)));
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(0,
rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet =
BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, capture_time_ms);
size_t packet_size = packet->size();
const int kStoredTimeInMs = 100;
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum))))));
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->set_allow_retransmission(true);
EXPECT_TRUE(
rtp_sender()->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
EXPECT_EQ(0, transport_.packets_sent());
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Process send bucket. Packet should now be sent.
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
webrtc::RTPHeader rtp_header;
transport_.last_sent_packet().GetHeader(&rtp_header);
// Verify transmission time offset.
EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset);
uint64_t expected_send_time =
ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
}
TEST_P(RtpSenderTest, TrafficSmoothingRetransmits) {
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)));
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(0,
rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet =
BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, capture_time_ms);
size_t packet_size = packet->size();
// Packet should be stored in a send bucket.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum))))));
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->set_allow_retransmission(true);
EXPECT_TRUE(
rtp_sender()->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
// Immediately process send bucket and send packet.
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
// Retransmit packet.
const int kStoredTimeInMs = 100;
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)));
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
packet->set_retransmitted_sequence_number(kSeqNum);
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum))))));
EXPECT_EQ(static_cast<int>(packet_size), rtp_sender()->ReSendPacket(kSeqNum));
EXPECT_EQ(1, transport_.packets_sent());
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Process send bucket. Packet should now be sent.
EXPECT_EQ(2, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
webrtc::RTPHeader rtp_header;
transport_.last_sent_packet().GetHeader(&rtp_header);
// Verify transmission time offset.
EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset);
uint64_t expected_send_time =
ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
}
// This test sends 1 regular video packet, then 4 padding packets, and then
// 1 more regular packet.
TEST_P(RtpSenderTest, SendPadding) {
// Make all (non-padding) packets go to send queue.
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(1 + 4 + 1);
uint16_t seq_num = kSeqNum;
uint32_t timestamp = kTimestamp;
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
size_t rtp_header_len = kRtpHeaderSize;
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
rtp_header_len += 4; // 4 bytes extension.
EXPECT_EQ(0,
rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
rtp_header_len += 4; // 4 bytes extension.
rtp_header_len += 4; // 4 extra bytes common to all extension headers.
webrtc::RTPHeader rtp_header;
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet =
BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
const uint32_t media_packet_timestamp = timestamp;
size_t packet_size = packet->size();
int total_packets_sent = 0;
const int kStoredTimeInMs = 100;
// Packet should be stored in a send bucket.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum))))));
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->set_allow_retransmission(true);
EXPECT_TRUE(
rtp_sender()->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
EXPECT_EQ(total_packets_sent, transport_.packets_sent());
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
++seq_num;
// Packet should now be sent. This test doesn't verify the regular video
// packet, since it is tested in another test.
EXPECT_EQ(++total_packets_sent, transport_.packets_sent());
timestamp += 90 * kStoredTimeInMs;
// Send padding 4 times, waiting 50 ms between each.
for (int i = 0; i < 4; ++i) {
const int kPaddingPeriodMs = 50;
const size_t kPaddingBytes = 100;
const size_t kMaxPaddingLength = 224; // Value taken from rtp_sender.cc.
// Padding will be forced to full packets.
EXPECT_EQ(kMaxPaddingLength, GenerateAndSendPadding(kPaddingBytes));
// Process send bucket. Padding should now be sent.
EXPECT_EQ(++total_packets_sent, transport_.packets_sent());
EXPECT_EQ(kMaxPaddingLength + rtp_header_len,
transport_.last_sent_packet().size());
transport_.last_sent_packet().GetHeader(&rtp_header);
EXPECT_EQ(kMaxPaddingLength, rtp_header.paddingLength);
// Verify sequence number and timestamp. The timestamp should be the same
// as the last media packet.
EXPECT_EQ(seq_num++, rtp_header.sequenceNumber);
EXPECT_EQ(media_packet_timestamp, rtp_header.timestamp);
// Verify transmission time offset.
int offset = timestamp - media_packet_timestamp;
EXPECT_EQ(offset, rtp_header.extension.transmissionTimeOffset);
uint64_t expected_send_time =
ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
fake_clock_.AdvanceTimeMilliseconds(kPaddingPeriodMs);
timestamp += 90 * kPaddingPeriodMs;
}
// Send a regular video packet again.
capture_time_ms = fake_clock_.TimeInMilliseconds();
packet = BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
packet_size = packet->size();
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->set_allow_retransmission(true);
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, seq_num))))));
EXPECT_TRUE(
rtp_sender()->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Process send bucket.
EXPECT_EQ(++total_packets_sent, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
transport_.last_sent_packet().GetHeader(&rtp_header);
// Verify sequence number and timestamp.
EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
EXPECT_EQ(timestamp, rtp_header.timestamp);
// Verify transmission time offset. This packet is sent without delay.
EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
uint64_t expected_send_time =
ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
}
TEST_P(RtpSenderTest, OnSendPacketUpdated) {
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
EXPECT_CALL(send_packet_observer_,
OnSendPacket(kTransportSequenceNumber, _, _))
.Times(1);
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum))))));
auto packet = SendGenericPacket();
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
}
TEST_P(RtpSenderTest, OnSendPacketNotUpdatedForRetransmits) {
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
EXPECT_CALL(send_packet_observer_, OnSendPacket(_, _, _)).Times(0);
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum))))));
auto packet = SendGenericPacket();
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_TRUE(transport_.last_options_.is_retransmit);
}
TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
const uint8_t kPayloadType = 127;
const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
uint8_t payload[] = {47, 11, 32, 93, 89};
// Send keyframe
RTPVideoHeader video_header;
video_header.frame_type = VideoFrameType::kVideoFrameKey;
ASSERT_TRUE(rtp_sender_video.SendVideo(kPayloadType, kCodecType, 1234, 4321,
payload, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
auto sent_payload = transport_.last_sent_packet().payload();
uint8_t generic_header = sent_payload[0];
EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(payload));
// Send delta frame
payload[0] = 13;
payload[1] = 42;
payload[4] = 13;
video_header.frame_type = VideoFrameType::kVideoFrameDelta;
ASSERT_TRUE(rtp_sender_video.SendVideo(kPayloadType, kCodecType, 1234, 4321,
payload, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
sent_payload = transport_.last_sent_packet().payload();
generic_header = sent_payload[0];
EXPECT_FALSE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(payload));
}
TEST_P(RtpSenderTestWithoutPacer, SendRawVideo) {
const uint8_t kPayloadType = 111;
const uint8_t payload[] = {11, 22, 33, 44, 55};
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
// Send a frame.
RTPVideoHeader video_header;
video_header.frame_type = VideoFrameType::kVideoFrameKey;
ASSERT_TRUE(rtp_sender_video.SendVideo(kPayloadType, absl::nullopt, 1234,
4321, payload, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
auto sent_payload = transport_.last_sent_packet().payload();
EXPECT_THAT(sent_payload, ElementsAreArray(payload));
}
TEST_P(RtpSenderTest, SendFlexfecPackets) {
constexpr uint32_t kTimestamp = 1234;
constexpr int kMediaPayloadType = 127;
constexpr VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
constexpr int kFlexfecPayloadType = 118;
const std::vector<RtpExtension> kNoRtpExtensions;
const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexFecSsrc, kSsrc, kNoMid,
kNoRtpExtensions, kNoRtpExtensionSizes,
nullptr /* rtp_state */, &fake_clock_);
// Reset |rtp_sender_| to use FlexFEC.
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.paced_sender = &mock_paced_sender_;
config.local_media_ssrc = kSsrc;
config.fec_generator = &flexfec_sender_;
config.event_log = &mock_rtc_event_log_;
config.send_packet_observer = &send_packet_observer_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
rtp_sender()->SetSequenceNumber(kSeqNum);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.fec_generator = &flexfec_sender;
video_config.fec_type = flexfec_sender.GetFecType();
video_config.fec_overhead_bytes = flexfec_sender.MaxPacketOverhead();
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
params.fec_rate = 15;
params.max_fec_frames = 1;
params.fec_mask_type = kFecMaskRandom;
flexfec_sender.SetProtectionParameters(params, params);
uint16_t flexfec_seq_num;
RTPVideoHeader video_header;
std::unique_ptr<RtpPacketToSend> media_packet;
std::unique_ptr<RtpPacketToSend> fec_packet;
EXPECT_CALL(mock_paced_sender_, EnqueuePackets)
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
for (auto& packet : packets) {
if (packet->packet_type() == RtpPacketMediaType::kVideo) {
EXPECT_EQ(packet->Ssrc(), kSsrc);
EXPECT_EQ(packet->SequenceNumber(), kSeqNum);
media_packet = std::move(packet);
} else {
EXPECT_EQ(packet->packet_type(),
RtpPacketMediaType::kForwardErrorCorrection);
EXPECT_EQ(packet->Ssrc(), kFlexFecSsrc);
fec_packet = std::move(packet);
}
}
});
video_header.frame_type = VideoFrameType::kVideoFrameKey;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kMediaPayloadType, kCodecType, kTimestamp,
fake_clock_.TimeInMilliseconds(), kPayloadData, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
ASSERT_TRUE(media_packet != nullptr);
ASSERT_TRUE(fec_packet != nullptr);
flexfec_seq_num = fec_packet->SequenceNumber();
rtp_egress()->SendPacket(media_packet.get(), PacedPacketInfo());
rtp_egress()->SendPacket(fec_packet.get(), PacedPacketInfo());
ASSERT_EQ(2, transport_.packets_sent());
const RtpPacketReceived& sent_media_packet = transport_.sent_packets_[0];
EXPECT_EQ(kMediaPayloadType, sent_media_packet.PayloadType());
EXPECT_EQ(kSeqNum, sent_media_packet.SequenceNumber());
EXPECT_EQ(kSsrc, sent_media_packet.Ssrc());
const RtpPacketReceived& sent_flexfec_packet = transport_.sent_packets_[1];
EXPECT_EQ(kFlexfecPayloadType, sent_flexfec_packet.PayloadType());
EXPECT_EQ(flexfec_seq_num, sent_flexfec_packet.SequenceNumber());
EXPECT_EQ(kFlexFecSsrc, sent_flexfec_packet.Ssrc());
}
TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
constexpr uint32_t kTimestamp = 1234;
constexpr int kMediaPayloadType = 127;
constexpr VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
constexpr int kFlexfecPayloadType = 118;
const std::vector<RtpExtension> kNoRtpExtensions;
const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexFecSsrc, kSsrc, kNoMid,
kNoRtpExtensions, kNoRtpExtensionSizes,
nullptr /* rtp_state */, &fake_clock_);
// Reset |rtp_sender_| to use FlexFEC.
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.fec_generator = &flexfec_sender;
config.event_log = &mock_rtc_event_log_;
config.send_packet_observer = &send_packet_observer_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
rtp_sender()->SetSequenceNumber(kSeqNum);
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.fec_generator = &flexfec_sender;
video_config.fec_type = flexfec_sender.GetFecType();
video_config.fec_overhead_bytes = flexfec_sender_.MaxPacketOverhead();
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
params.fec_rate = 15;
params.max_fec_frames = 1;
params.fec_mask_type = kFecMaskRandom;
flexfec_sender.SetProtectionParameters(params, params);
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(2);
RTPVideoHeader video_header;
video_header.frame_type = VideoFrameType::kVideoFrameKey;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kMediaPayloadType, kCodecType, kTimestamp,
fake_clock_.TimeInMilliseconds(), kPayloadData, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
ASSERT_EQ(2, transport_.packets_sent());
const RtpPacketReceived& media_packet = transport_.sent_packets_[0];
EXPECT_EQ(kMediaPayloadType, media_packet.PayloadType());
EXPECT_EQ(kSsrc, media_packet.Ssrc());
const RtpPacketReceived& flexfec_packet = transport_.sent_packets_[1];
EXPECT_EQ(kFlexfecPayloadType, flexfec_packet.PayloadType());
EXPECT_EQ(kFlexFecSsrc, flexfec_packet.Ssrc());
}
// Test that the MID header extension is included on sent packets when
// configured.
TEST_P(RtpSenderTestWithoutPacer, MidIncludedOnSentPackets) {
const char kMid[] = "mid";
EnableMidSending(kMid);
// Send a couple packets.
SendGenericPacket();
SendGenericPacket();
// Expect both packets to have the MID set.
ASSERT_EQ(2u, transport_.sent_packets_.size());
for (const RtpPacketReceived& packet : transport_.sent_packets_) {
std::string mid;
ASSERT_TRUE(packet.GetExtension<RtpMid>(&mid));
EXPECT_EQ(kMid, mid);
}
}
TEST_P(RtpSenderTestWithoutPacer, RidIncludedOnSentPackets) {
const char kRid[] = "f";
EnableRidSending(kRid);
SendGenericPacket();
ASSERT_EQ(1u, transport_.sent_packets_.size());
const RtpPacketReceived& packet = transport_.sent_packets_[0];
std::string rid;
ASSERT_TRUE(packet.GetExtension<RtpStreamId>(&rid));
EXPECT_EQ(kRid, rid);
}
TEST_P(RtpSenderTestWithoutPacer, RidIncludedOnRtxSentPackets) {
const char kRid[] = "f";
EnableRtx();
EnableRidSending(kRid);
SendGenericPacket();
ASSERT_EQ(1u, transport_.sent_packets_.size());
const RtpPacketReceived& packet = transport_.sent_packets_[0];
std::string rid;
ASSERT_TRUE(packet.GetExtension<RtpStreamId>(&rid));
EXPECT_EQ(kRid, rid);
rid = kNoRid;
EXPECT_FALSE(packet.HasExtension<RepairedRtpStreamId>());
uint16_t packet_id = packet.SequenceNumber();
rtp_sender()->ReSendPacket(packet_id);
ASSERT_EQ(2u, transport_.sent_packets_.size());
const RtpPacketReceived& rtx_packet = transport_.sent_packets_[1];
ASSERT_TRUE(rtx_packet.GetExtension<RepairedRtpStreamId>(&rid));
EXPECT_EQ(kRid, rid);
EXPECT_FALSE(rtx_packet.HasExtension<RtpStreamId>());
}
TEST_P(RtpSenderTestWithoutPacer, MidAndRidNotIncludedOnSentPacketsAfterAck) {
const char kMid[] = "mid";
const char kRid[] = "f";
EnableMidSending(kMid);
EnableRidSending(kRid);
// This first packet should include both MID and RID.
auto first_built_packet = SendGenericPacket();
rtp_sender()->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
// The second packet should include neither since an ack was received.
SendGenericPacket();
ASSERT_EQ(2u, transport_.sent_packets_.size());
const RtpPacketReceived& first_packet = transport_.sent_packets_[0];
std::string mid, rid;
ASSERT_TRUE(first_packet.GetExtension<RtpMid>(&mid));
EXPECT_EQ(kMid, mid);
ASSERT_TRUE(first_packet.GetExtension<RtpStreamId>(&rid));
EXPECT_EQ(kRid, rid);
const RtpPacketReceived& second_packet = transport_.sent_packets_[1];
EXPECT_FALSE(second_packet.HasExtension<RtpMid>());
EXPECT_FALSE(second_packet.HasExtension<RtpStreamId>());
}
TEST_P(RtpSenderTestWithoutPacer,
MidAndRidAlwaysIncludedOnSentPacketsWhenConfigured) {
SetUpRtpSender(false, false, /*always_send_mid_and_rid=*/true);
const char kMid[] = "mid";
const char kRid[] = "f";
EnableMidSending(kMid);
EnableRidSending(kRid);
// Send two media packets: one before and one after the ack.
auto first_packet = SendGenericPacket();
rtp_sender()->OnReceivedAckOnSsrc(first_packet->SequenceNumber());
SendGenericPacket();
// Due to the configuration, both sent packets should contain MID and RID.
ASSERT_EQ(2u, transport_.sent_packets_.size());
for (const RtpPacketReceived& packet : transport_.sent_packets_) {
EXPECT_EQ(packet.GetExtension<RtpMid>(), kMid);
EXPECT_EQ(packet.GetExtension<RtpStreamId>(), kRid);
}
}
// Test that the first RTX packet includes both MID and RRID even if the packet
// being retransmitted did not have MID or RID. The MID and RID are needed on
// the first packets for a given SSRC, and RTX packets are sent on a separate
// SSRC.
TEST_P(RtpSenderTestWithoutPacer, MidAndRidIncludedOnFirstRtxPacket) {
const char kMid[] = "mid";
const char kRid[] = "f";
EnableRtx();
EnableMidSending(kMid);
EnableRidSending(kRid);
// This first packet will include both MID and RID.
auto first_built_packet = SendGenericPacket();
rtp_sender()->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
// The second packet will include neither since an ack was received.
auto second_built_packet = SendGenericPacket();
// The first RTX packet should include MID and RRID.
ASSERT_LT(0,
rtp_sender()->ReSendPacket(second_built_packet->SequenceNumber()));
ASSERT_EQ(3u, transport_.sent_packets_.size());
const RtpPacketReceived& rtx_packet = transport_.sent_packets_[2];
std::string mid, rrid;
ASSERT_TRUE(rtx_packet.GetExtension<RtpMid>(&mid));
EXPECT_EQ(kMid, mid);
ASSERT_TRUE(rtx_packet.GetExtension<RepairedRtpStreamId>(&rrid));
EXPECT_EQ(kRid, rrid);
}
// Test that the RTX packets sent after receving an ACK on the RTX SSRC does
// not include either MID or RRID even if the packet being retransmitted did
// had a MID or RID.
TEST_P(RtpSenderTestWithoutPacer, MidAndRidNotIncludedOnRtxPacketsAfterAck) {
const char kMid[] = "mid";
const char kRid[] = "f";
EnableRtx();
EnableMidSending(kMid);
EnableRidSending(kRid);
// This first packet will include both MID and RID.
auto first_built_packet = SendGenericPacket();
rtp_sender()->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
// The second packet will include neither since an ack was received.
auto second_built_packet = SendGenericPacket();
// The first RTX packet will include MID and RRID.
ASSERT_LT(0,
rtp_sender()->ReSendPacket(second_built_packet->SequenceNumber()));
ASSERT_EQ(3u, transport_.sent_packets_.size());
const RtpPacketReceived& first_rtx_packet = transport_.sent_packets_[2];
rtp_sender()->OnReceivedAckOnRtxSsrc(first_rtx_packet.SequenceNumber());
// The second and third RTX packets should not include MID nor RRID.
ASSERT_LT(0,
rtp_sender()->ReSendPacket(first_built_packet->SequenceNumber()));
ASSERT_LT(0,
rtp_sender()->ReSendPacket(second_built_packet->SequenceNumber()));
ASSERT_EQ(5u, transport_.sent_packets_.size());
const RtpPacketReceived& second_rtx_packet = transport_.sent_packets_[3];
EXPECT_FALSE(second_rtx_packet.HasExtension<RtpMid>());
EXPECT_FALSE(second_rtx_packet.HasExtension<RepairedRtpStreamId>());
const RtpPacketReceived& third_rtx_packet = transport_.sent_packets_[4];
EXPECT_FALSE(third_rtx_packet.HasExtension<RtpMid>());
EXPECT_FALSE(third_rtx_packet.HasExtension<RepairedRtpStreamId>());
}
TEST_P(RtpSenderTestWithoutPacer,
MidAndRidAlwaysIncludedOnRtxPacketsWhenConfigured) {
SetUpRtpSender(false, false, /*always_send_mid_and_rid=*/true);
const char kMid[] = "mid";
const char kRid[] = "f";
EnableRtx();
EnableMidSending(kMid);
EnableRidSending(kRid);
// Send two media packets: one before and one after the ack.
auto media_packet1 = SendGenericPacket();
rtp_sender()->OnReceivedAckOnSsrc(media_packet1->SequenceNumber());
auto media_packet2 = SendGenericPacket();
// Send three RTX packets with different combinations of orders w.r.t. the
// media and RTX acks.
ASSERT_LT(0, rtp_sender()->ReSendPacket(media_packet2->SequenceNumber()));
ASSERT_EQ(3u, transport_.sent_packets_.size());
rtp_sender()->OnReceivedAckOnRtxSsrc(
transport_.sent_packets_[2].SequenceNumber());
ASSERT_LT(0, rtp_sender()->ReSendPacket(media_packet1->SequenceNumber()));
ASSERT_LT(0, rtp_sender()->ReSendPacket(media_packet2->SequenceNumber()));
// Due to the configuration, all sent packets should contain MID
// and either RID (media) or RRID (RTX).
ASSERT_EQ(5u, transport_.sent_packets_.size());
for (const auto& packet : transport_.sent_packets_) {
EXPECT_EQ(packet.GetExtension<RtpMid>(), kMid);
}
for (size_t i = 0; i < 2; ++i) {
const RtpPacketReceived& packet = transport_.sent_packets_[i];
EXPECT_EQ(packet.GetExtension<RtpStreamId>(), kRid);
}
for (size_t i = 2; i < transport_.sent_packets_.size(); ++i) {
const RtpPacketReceived& packet = transport_.sent_packets_[i];
EXPECT_EQ(packet.GetExtension<RepairedRtpStreamId>(), kRid);
}
}
// Test that if the RtpState indicates an ACK has been received on that SSRC
// then neither the MID nor RID header extensions will be sent.
TEST_P(RtpSenderTestWithoutPacer,
MidAndRidNotIncludedOnSentPacketsAfterRtpStateRestored) {
const char kMid[] = "mid";
const char kRid[] = "f";
EnableMidSending(kMid);
EnableRidSending(kRid);
RtpState state = rtp_sender()->GetRtpState();
EXPECT_FALSE(state.ssrc_has_acked);
state.ssrc_has_acked = true;
rtp_sender()->SetRtpState(state);
SendGenericPacket();
ASSERT_EQ(1u, transport_.sent_packets_.size());
const RtpPacketReceived& packet = transport_.sent_packets_[0];
EXPECT_FALSE(packet.HasExtension<RtpMid>());
EXPECT_FALSE(packet.HasExtension<RtpStreamId>());
}
// Test that if the RTX RtpState indicates an ACK has been received on that
// RTX SSRC then neither the MID nor RRID header extensions will be sent on
// RTX packets.
TEST_P(RtpSenderTestWithoutPacer,
MidAndRridNotIncludedOnRtxPacketsAfterRtpStateRestored) {
const char kMid[] = "mid";
const char kRid[] = "f";
EnableRtx();
EnableMidSending(kMid);
EnableRidSending(kRid);
RtpState rtx_state = rtp_sender()->GetRtxRtpState();
EXPECT_FALSE(rtx_state.ssrc_has_acked);
rtx_state.ssrc_has_acked = true;
rtp_sender()->SetRtxRtpState(rtx_state);
auto built_packet = SendGenericPacket();
ASSERT_LT(0, rtp_sender()->ReSendPacket(built_packet->SequenceNumber()));
ASSERT_EQ(2u, transport_.sent_packets_.size());
const RtpPacketReceived& rtx_packet = transport_.sent_packets_[1];
EXPECT_FALSE(rtx_packet.HasExtension<RtpMid>());
EXPECT_FALSE(rtx_packet.HasExtension<RepairedRtpStreamId>());
}
TEST_P(RtpSenderTest, FecOverheadRate) {
constexpr uint32_t kTimestamp = 1234;
constexpr int kMediaPayloadType = 127;
constexpr VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
constexpr int kFlexfecPayloadType = 118;
const std::vector<RtpExtension> kNoRtpExtensions;
const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexFecSsrc, kSsrc, kNoMid,
kNoRtpExtensions, kNoRtpExtensionSizes,
nullptr /* rtp_state */, &fake_clock_);
// Reset |rtp_sender_| to use FlexFEC.
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.paced_sender = &mock_paced_sender_;
config.local_media_ssrc = kSsrc;
config.fec_generator = &flexfec_sender;
config.event_log = &mock_rtc_event_log_;
config.send_packet_observer = &send_packet_observer_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
rtp_sender()->SetSequenceNumber(kSeqNum);
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.fec_generator = &flexfec_sender;
video_config.fec_type = flexfec_sender.GetFecType();
video_config.fec_overhead_bytes = flexfec_sender.MaxPacketOverhead();
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
params.fec_rate = 15;
params.max_fec_frames = 1;
params.fec_mask_type = kFecMaskRandom;
flexfec_sender.SetProtectionParameters(params, params);
constexpr size_t kNumMediaPackets = 10;
constexpr size_t kNumFecPackets = kNumMediaPackets;
constexpr int64_t kTimeBetweenPacketsMs = 10;
EXPECT_CALL(mock_paced_sender_, EnqueuePackets).Times(kNumMediaPackets);
for (size_t i = 0; i < kNumMediaPackets; ++i) {
RTPVideoHeader video_header;
video_header.frame_type = VideoFrameType::kVideoFrameKey;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kMediaPayloadType, kCodecType, kTimestamp,
fake_clock_.TimeInMilliseconds(), kPayloadData, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
fake_clock_.AdvanceTimeMilliseconds(kTimeBetweenPacketsMs);
}
constexpr size_t kRtpHeaderLength = 12;
constexpr size_t kFlexfecHeaderLength = 20;
constexpr size_t kGenericCodecHeaderLength = 1;
constexpr size_t kPayloadLength = sizeof(kPayloadData);
constexpr size_t kPacketLength = kRtpHeaderLength + kFlexfecHeaderLength +
kGenericCodecHeaderLength + kPayloadLength;
EXPECT_NEAR(kNumFecPackets * kPacketLength * 8 /
(kNumFecPackets * kTimeBetweenPacketsMs / 1000.0f),
flexfec_sender.CurrentFecRate().bps<double>(), 500);
}
TEST_P(RtpSenderTest, BitrateCallbacks) {
class TestCallback : public BitrateStatisticsObserver {
public:
TestCallback()
: BitrateStatisticsObserver(),
num_calls_(0),
ssrc_(0),
total_bitrate_(0),
retransmit_bitrate_(0) {}
~TestCallback() override = default;
void Notify(uint32_t total_bitrate,
uint32_t retransmit_bitrate,
uint32_t ssrc) override {
++num_calls_;
ssrc_ = ssrc;
total_bitrate_ = total_bitrate;
retransmit_bitrate_ = retransmit_bitrate;
}
uint32_t num_calls_;
uint32_t ssrc_;
uint32_t total_bitrate_;
uint32_t retransmit_bitrate_;
} callback;
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.send_bitrate_observer = &callback;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
const uint8_t kPayloadType = 127;
// Simulate kNumPackets sent with kPacketInterval ms intervals, with the
// number of packets selected so that we fill (but don't overflow) the one
// second averaging window.
const uint32_t kWindowSizeMs = 1000;
const uint32_t kPacketInterval = 20;
const uint32_t kNumPackets =
(kWindowSizeMs - kPacketInterval) / kPacketInterval;
// Overhead = 12 bytes RTP header + 1 byte generic header.
const uint32_t kPacketOverhead = 13;
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
uint32_t ssrc = rtp_sender()->SSRC();
// Initial process call so we get a new time window.
rtp_egress()->ProcessBitrateAndNotifyObservers();
// Send a few frames.
RTPVideoHeader video_header;
for (uint32_t i = 0; i < kNumPackets; ++i) {
video_header.frame_type = VideoFrameType::kVideoFrameKey;
ASSERT_TRUE(rtp_sender_video.SendVideo(
kPayloadType, kCodecType, 1234, 4321, payload, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
}
rtp_egress()->ProcessBitrateAndNotifyObservers();
// We get one call for every stats updated, thus two calls since both the
// stream stats and the retransmit stats are updated once.
EXPECT_EQ(2u, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
const uint32_t kTotalPacketSize = kPacketOverhead + sizeof(payload);
// Bitrate measured over delta between last and first timestamp, plus one.
const uint32_t kExpectedWindowMs = kNumPackets * kPacketInterval + 1;
const uint32_t kExpectedBitsAccumulated = kTotalPacketSize * kNumPackets * 8;
const uint32_t kExpectedRateBps =
(kExpectedBitsAccumulated * 1000 + (kExpectedWindowMs / 2)) /
kExpectedWindowMs;
EXPECT_EQ(kExpectedRateBps, callback.total_bitrate_);
}
TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
const uint8_t kPayloadType = 127;
const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
uint32_t ssrc = rtp_sender()->SSRC();
// Send a frame.
RTPVideoHeader video_header;
video_header.frame_type = VideoFrameType::kVideoFrameKey;
ASSERT_TRUE(rtp_sender_video.SendVideo(kPayloadType, kCodecType, 1234, 4321,
payload, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
StreamDataCounters expected;
expected.transmitted.payload_bytes = 6;
expected.transmitted.header_bytes = 12;
expected.transmitted.padding_bytes = 0;
expected.transmitted.packets = 1;
expected.retransmitted.payload_bytes = 0;
expected.retransmitted.header_bytes = 0;
expected.retransmitted.padding_bytes = 0;
expected.retransmitted.packets = 0;
expected.fec.packets = 0;
rtp_stats_callback_.Matches(ssrc, expected);
// Retransmit a frame.
uint16_t seqno = rtp_sender()->SequenceNumber() - 1;
rtp_sender()->ReSendPacket(seqno);
expected.transmitted.payload_bytes = 12;
expected.transmitted.header_bytes = 24;
expected.transmitted.packets = 2;
expected.retransmitted.payload_bytes = 6;
expected.retransmitted.header_bytes = 12;
expected.retransmitted.padding_bytes = 0;
expected.retransmitted.packets = 1;
rtp_stats_callback_.Matches(ssrc, expected);
// Send padding.
GenerateAndSendPadding(kMaxPaddingSize);
expected.transmitted.payload_bytes = 12;
expected.transmitted.header_bytes = 36;
expected.transmitted.padding_bytes = kMaxPaddingSize;
expected.transmitted.packets = 3;
rtp_stats_callback_.Matches(ssrc, expected);
}
TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacksUlpfec) {
const uint8_t kRedPayloadType = 96;
const uint8_t kUlpfecPayloadType = 97;
const uint8_t kPayloadType = 127;
const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
FieldTrialBasedConfig field_trials;
UlpfecGenerator ulpfec_generator(kRedPayloadType, kUlpfecPayloadType,
&fake_clock_);
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.field_trials = &field_trials;
video_config.red_payload_type = kRedPayloadType;
video_config.fec_generator = &ulpfec_generator;
video_config.fec_type = ulpfec_generator.GetFecType();
video_config.fec_overhead_bytes = ulpfec_generator.MaxPacketOverhead();
RTPSenderVideo rtp_sender_video(video_config);
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
uint32_t ssrc = rtp_sender()->SSRC();
RTPVideoHeader video_header;
StreamDataCounters expected;
// Send ULPFEC.
FecProtectionParams fec_params;
fec_params.fec_mask_type = kFecMaskRandom;
fec_params.fec_rate = 1;
fec_params.max_fec_frames = 1;
ulpfec_generator.SetProtectionParameters(fec_params, fec_params);
video_header.frame_type = VideoFrameType::kVideoFrameDelta;
ASSERT_TRUE(rtp_sender_video.SendVideo(kPayloadType, kCodecType, 1234, 4321,
payload, nullptr, video_header,
kDefaultExpectedRetransmissionTimeMs));
expected.transmitted.payload_bytes = 28;
expected.transmitted.header_bytes = 24;
expected.transmitted.packets = 2;
expected.fec.packets = 1;
rtp_stats_callback_.Matches(ssrc, expected);
}
TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
// XXX const char* kPayloadName = "GENERIC";
const uint8_t kPayloadType = 127;
rtp_sender()->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
SendGenericPacket();
// Will send 2 full-size padding packets.
GenerateAndSendPadding(1);
GenerateAndSendPadding(1);
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
rtp_egress()->GetDataCounters(&rtp_stats, &rtx_stats);
// Payload
EXPECT_GT(rtp_stats.first_packet_time_ms, -1);
EXPECT_EQ(rtp_stats.transmitted.payload_bytes, sizeof(kPayloadData));
EXPECT_EQ(rtp_stats.transmitted.header_bytes, 12u);
EXPECT_EQ(rtp_stats.transmitted.padding_bytes, 0u);
EXPECT_EQ(rtx_stats.transmitted.payload_bytes, 0u);
EXPECT_EQ(rtx_stats.transmitted.header_bytes, 24u);
EXPECT_EQ(rtx_stats.transmitted.padding_bytes, 2 * kMaxPaddingSize);
EXPECT_EQ(rtp_stats.transmitted.TotalBytes(),
rtp_stats.transmitted.payload_bytes +
rtp_stats.transmitted.header_bytes +
rtp_stats.transmitted.padding_bytes);
EXPECT_EQ(rtx_stats.transmitted.TotalBytes(),
rtx_stats.transmitted.payload_bytes +
rtx_stats.transmitted.header_bytes +
rtx_stats.transmitted.padding_bytes);
EXPECT_EQ(
transport_.total_bytes_sent_,
rtp_stats.transmitted.TotalBytes() + rtx_stats.transmitted.TotalBytes());
}
TEST_P(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
const int32_t kPacketSize = 1400;
const int32_t kNumPackets = 30;
retransmission_rate_limiter_.SetMaxRate(kPacketSize * kNumPackets * 8);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, kNumPackets);
const uint16_t kStartSequenceNumber = rtp_sender()->SequenceNumber();
std::vector<uint16_t> sequence_numbers;
for (int32_t i = 0; i < kNumPackets; ++i) {
sequence_numbers.push_back(kStartSequenceNumber + i);
fake_clock_.AdvanceTimeMilliseconds(1);
SendPacket(fake_clock_.TimeInMilliseconds(), kPacketSize);
}
EXPECT_EQ(kNumPackets, transport_.packets_sent());
fake_clock_.AdvanceTimeMilliseconds(1000 - kNumPackets);
// Resending should work - brings the bandwidth up to the limit.
// NACK bitrate is capped to the same bitrate as the encoder, since the max
// protection overhead is 50% (see MediaOptimization::SetTargetRates).
rtp_sender()->OnReceivedNack(sequence_numbers, 0);
EXPECT_EQ(kNumPackets * 2, transport_.packets_sent());
// Must be at least 5ms in between retransmission attempts.
fake_clock_.AdvanceTimeMilliseconds(5);
// Resending should not work, bandwidth exceeded.
rtp_sender()->OnReceivedNack(sequence_numbers, 0);
EXPECT_EQ(kNumPackets * 2, transport_.packets_sent());
}
TEST_P(RtpSenderTest, UpdatingCsrcsUpdatedOverhead) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
// Adding two csrcs adds 2*4 bytes to the header.
rtp_sender()->SetCsrcs({1, 2});
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 20u);
}
TEST_P(RtpSenderTest, OnOverheadChanged) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
// TransmissionTimeOffset extension has a size of 3B, but with the addition
// of header index and rounding to 4 byte boundary we end up with 20B total.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 20u);
}
TEST_P(RtpSenderTest, CountMidOnlyUntilAcked) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionMid, kMidExtensionId);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionRtpStreamId,
kRidExtensionId);
// Counted only if set.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
rtp_sender()->SetMid("foo");
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 36u);
rtp_sender()->SetRid("bar");
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 52u);
// Ack received, mid/rid no longer sent.
rtp_sender()->OnReceivedAckOnSsrc(0);
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
}
TEST_P(RtpSenderTest, DontCountVolatileExtensionsIntoOverhead) {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionInbandComfortNoise, 1);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteCaptureTime, 2);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionVideoRotation, 3);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionPlayoutDelay, 4);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionVideoContentType, 5);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionVideoTiming, 6);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionRepairedRtpStreamId, 7);
rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionColorSpace, 8);
// Still only 12B counted since can't count on above being sent.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
}
TEST_P(RtpSenderTest, SendPacketMatchesVideo) {
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_packet_type(RtpPacketMediaType::kVideo);
// Verify sent with correct SSRC.
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetSsrc(kSsrc);
packet->set_packet_type(RtpPacketMediaType::kVideo);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(transport_.packets_sent(), 1);
}
TEST_P(RtpSenderTest, SendPacketMatchesAudio) {
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_packet_type(RtpPacketMediaType::kAudio);
// Verify sent with correct SSRC.
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetSsrc(kSsrc);
packet->set_packet_type(RtpPacketMediaType::kAudio);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(transport_.packets_sent(), 1);
}
TEST_P(RtpSenderTest, SendPacketMatchesRetransmissions) {
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
// Verify sent with correct SSRC (non-RTX).
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetSsrc(kSsrc);
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(transport_.packets_sent(), 1);
// RTX retransmission.
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetSsrc(kRtxSsrc);
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(transport_.packets_sent(), 2);
}
TEST_P(RtpSenderTest, SendPacketMatchesPadding) {
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_packet_type(RtpPacketMediaType::kPadding);
// Verify sent with correct SSRC (non-RTX).
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetSsrc(kSsrc);
packet->set_packet_type(RtpPacketMediaType::kPadding);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(transport_.packets_sent(), 1);
// RTX padding.
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetSsrc(kRtxSsrc);
packet->set_packet_type(RtpPacketMediaType::kPadding);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(transport_.packets_sent(), 2);
}
TEST_P(RtpSenderTest, SendPacketMatchesFlexfec) {
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
// Verify sent with correct SSRC.
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetSsrc(kFlexFecSsrc);
packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(transport_.packets_sent(), 1);
}
TEST_P(RtpSenderTest, SendPacketMatchesUlpfec) {
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
// Verify sent with correct SSRC.
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetSsrc(kSsrc);
packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(transport_.packets_sent(), 1);
}
TEST_P(RtpSenderTest, SendPacketHandlesRetransmissionHistory) {
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
// Build a media packet and send it.
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
const uint16_t media_sequence_number = packet->SequenceNumber();
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->set_allow_retransmission(true);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Simulate retransmission request.
fake_clock_.AdvanceTimeMilliseconds(30);
EXPECT_GT(rtp_sender()->ReSendPacket(media_sequence_number), 0);
// Packet already pending, retransmission not allowed.
fake_clock_.AdvanceTimeMilliseconds(30);
EXPECT_EQ(rtp_sender()->ReSendPacket(media_sequence_number), 0);
// Packet exiting pacer, mark as not longer pending.
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
EXPECT_NE(packet->SequenceNumber(), media_sequence_number);
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
packet->SetSsrc(kRtxSsrc);
packet->set_retransmitted_sequence_number(media_sequence_number);
packet->set_allow_retransmission(false);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Retransmissions allowed again.
fake_clock_.AdvanceTimeMilliseconds(30);
EXPECT_GT(rtp_sender()->ReSendPacket(media_sequence_number), 0);
// Retransmission of RTX packet should not be allowed.
EXPECT_EQ(rtp_sender()->ReSendPacket(packet->SequenceNumber()), 0);
}
TEST_P(RtpSenderTest, SendPacketUpdatesExtensions) {
ASSERT_EQ(rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId),
0);
ASSERT_EQ(rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId),
0);
ASSERT_EQ(rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionVideoTiming,
kVideoTimingExtensionId),
0);
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_packetization_finish_time_ms(fake_clock_.TimeInMilliseconds());
const int32_t kDiffMs = 10;
fake_clock_.AdvanceTimeMilliseconds(kDiffMs);
packet->set_packet_type(RtpPacketMediaType::kVideo);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
const RtpPacketReceived& received_packet = transport_.last_sent_packet();
EXPECT_EQ(received_packet.GetExtension<TransmissionOffset>(), kDiffMs * 90);
EXPECT_EQ(received_packet.GetExtension<AbsoluteSendTime>(),
AbsoluteSendTime::MsTo24Bits(fake_clock_.TimeInMilliseconds()));
VideoSendTiming timing;
EXPECT_TRUE(received_packet.GetExtension<VideoTimingExtension>(&timing));
EXPECT_EQ(timing.pacer_exit_delta_ms, kDiffMs);
}
TEST_P(RtpSenderTest, SendPacketSetsPacketOptions) {
const uint16_t kPacketId = 42;
ASSERT_EQ(rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId),
0);
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetExtension<TransportSequenceNumber>(kPacketId);
packet->set_packet_type(RtpPacketMediaType::kVideo);
EXPECT_CALL(send_packet_observer_, OnSendPacket);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_EQ(transport_.last_options_.packet_id, kPacketId);
EXPECT_TRUE(transport_.last_options_.included_in_allocation);
EXPECT_TRUE(transport_.last_options_.included_in_feedback);
EXPECT_FALSE(transport_.last_options_.is_retransmit);
// Send another packet as retransmission, verify options are populated.
packet = BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->SetExtension<TransportSequenceNumber>(kPacketId + 1);
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
EXPECT_TRUE(transport_.last_options_.is_retransmit);
}
TEST_P(RtpSenderTest, SendPacketUpdatesStats) {
const size_t kPayloadSize = 1000;
StrictMock<MockSendSideDelayObserver> send_side_delay_observer;
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.rtx_send_ssrc = kRtxSsrc;
config.fec_generator = &flexfec_sender_;
config.send_side_delay_observer = &send_side_delay_observer;
config.event_log = &mock_rtc_event_log_;
config.send_packet_observer = &send_packet_observer_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
const int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
std::unique_ptr<RtpPacketToSend> video_packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
video_packet->set_packet_type(RtpPacketMediaType::kVideo);
video_packet->SetPayloadSize(kPayloadSize);
video_packet->SetExtension<TransportSequenceNumber>(1);
std::unique_ptr<RtpPacketToSend> rtx_packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
rtx_packet->SetSsrc(kRtxSsrc);
rtx_packet->set_packet_type(RtpPacketMediaType::kRetransmission);
rtx_packet->SetPayloadSize(kPayloadSize);
rtx_packet->SetExtension<TransportSequenceNumber>(2);
std::unique_ptr<RtpPacketToSend> fec_packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
fec_packet->SetSsrc(kFlexFecSsrc);
fec_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
fec_packet->SetPayloadSize(kPayloadSize);
fec_packet->SetExtension<TransportSequenceNumber>(3);
const int64_t kDiffMs = 25;
fake_clock_.AdvanceTimeMilliseconds(kDiffMs);
EXPECT_CALL(send_side_delay_observer,
SendSideDelayUpdated(kDiffMs, kDiffMs, kDiffMs, kSsrc));
EXPECT_CALL(
send_side_delay_observer,
SendSideDelayUpdated(kDiffMs, kDiffMs, 2 * kDiffMs, kFlexFecSsrc));
EXPECT_CALL(send_packet_observer_, OnSendPacket(1, capture_time_ms, kSsrc));
rtp_egress()->SendPacket(video_packet.get(), PacedPacketInfo());
// Send packet observer not called for padding/retransmissions.
EXPECT_CALL(send_packet_observer_, OnSendPacket(2, _, _)).Times(0);
rtp_egress()->SendPacket(rtx_packet.get(), PacedPacketInfo());
EXPECT_CALL(send_packet_observer_,
OnSendPacket(3, capture_time_ms, kFlexFecSsrc));
rtp_egress()->SendPacket(fec_packet.get(), PacedPacketInfo());
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
rtp_egress()->GetDataCounters(&rtp_stats, &rtx_stats);
EXPECT_EQ(rtp_stats.transmitted.packets, 2u);
EXPECT_EQ(rtp_stats.fec.packets, 1u);
EXPECT_EQ(rtx_stats.retransmitted.packets, 1u);
}
TEST_P(RtpSenderTest, GeneratedPaddingHasBweExtensions) {
// Min requested size in order to use RTX payload.
const size_t kMinPaddingSize = 50;
rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
ASSERT_EQ(0,
rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
// Send a payload packet first, to enable padding and populate the packet
// history.
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kMinPaddingSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Generate a plain padding packet, check that extensions are registered.
std::vector<std::unique_ptr<RtpPacketToSend>> generated_packets =
rtp_sender()->GeneratePadding(/*target_size_bytes=*/1, true);
ASSERT_THAT(generated_packets, SizeIs(1));
auto& plain_padding = generated_packets.front();
EXPECT_GT(plain_padding->padding_size(), 0u);
EXPECT_TRUE(plain_padding->HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(plain_padding->HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(plain_padding->HasExtension<TransmissionOffset>());
// Verify all header extensions have been written.
rtp_egress()->SendPacket(plain_padding.get(), PacedPacketInfo());
const auto& sent_plain_padding = transport_.last_sent_packet();
EXPECT_TRUE(sent_plain_padding.HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(sent_plain_padding.HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(sent_plain_padding.HasExtension<TransmissionOffset>());
webrtc::RTPHeader rtp_header;
sent_plain_padding.GetHeader(&rtp_header);
EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
// Generate a payload padding packets, check that extensions are registered.
generated_packets = rtp_sender()->GeneratePadding(kMinPaddingSize, true);
ASSERT_EQ(generated_packets.size(), 1u);
auto& payload_padding = generated_packets.front();
EXPECT_EQ(payload_padding->padding_size(), 0u);
EXPECT_TRUE(payload_padding->HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(payload_padding->HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(payload_padding->HasExtension<TransmissionOffset>());
// Verify all header extensions have been written.
rtp_egress()->SendPacket(payload_padding.get(), PacedPacketInfo());
const auto& sent_payload_padding = transport_.last_sent_packet();
EXPECT_TRUE(sent_payload_padding.HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(sent_payload_padding.HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(sent_payload_padding.HasExtension<TransmissionOffset>());
sent_payload_padding.GetHeader(&rtp_header);
EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
}
TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
// Min requested size in order to use RTX payload.
const size_t kMinPaddingSize = 50;
rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
const size_t kPayloadPacketSize = kMinPaddingSize;
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
// Send a dummy video packet so it ends up in the packet history.
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Generated padding has large enough budget that the video packet should be
// retransmitted as padding.
std::vector<std::unique_ptr<RtpPacketToSend>> generated_packets =
rtp_sender()->GeneratePadding(kMinPaddingSize, true);
ASSERT_EQ(generated_packets.size(), 1u);
auto& padding_packet = generated_packets.front();
EXPECT_EQ(padding_packet->packet_type(), RtpPacketMediaType::kPadding);
EXPECT_EQ(padding_packet->Ssrc(), kRtxSsrc);
EXPECT_EQ(padding_packet->payload_size(),
kPayloadPacketSize + kRtxHeaderSize);
// Not enough budged for payload padding, use plain padding instead.
const size_t kPaddingBytesRequested = kMinPaddingSize - 1;
size_t padding_bytes_generated = 0;
generated_packets =
rtp_sender()->GeneratePadding(kPaddingBytesRequested, true);
EXPECT_EQ(generated_packets.size(), 1u);
for (auto& packet : generated_packets) {
EXPECT_EQ(packet->packet_type(), RtpPacketMediaType::kPadding);
EXPECT_EQ(packet->Ssrc(), kRtxSsrc);
EXPECT_EQ(packet->payload_size(), 0u);
EXPECT_GT(packet->padding_size(), 0u);
padding_bytes_generated += packet->padding_size();
}
EXPECT_EQ(padding_bytes_generated, kMaxPaddingSize);
}
TEST_P(RtpSenderTest, LimitsPayloadPaddingSize) {
// Limit RTX payload padding to 2x target size.
const double kFactor = 2.0;
field_trials_.SetMaxPaddingFactor(kFactor);
SetUpRtpSender(true, false, false);
rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
// Send a dummy video packet so it ends up in the packet history.
const size_t kPayloadPacketSize = 1234u;
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Smallest target size that will result in the sent packet being returned as
// padding.
const size_t kMinTargerSizeForPayload =
(kPayloadPacketSize + kRtxHeaderSize) / kFactor;
// Generated padding has large enough budget that the video packet should be
// retransmitted as padding.
EXPECT_THAT(
rtp_sender()->GeneratePadding(kMinTargerSizeForPayload, true),
AllOf(Not(IsEmpty()),
Each(Pointee(Property(&RtpPacketToSend::padding_size, Eq(0u))))));
// If payload padding is > 2x requested size, plain padding is returned
// instead.
EXPECT_THAT(
rtp_sender()->GeneratePadding(kMinTargerSizeForPayload - 1, true),
AllOf(Not(IsEmpty()),
Each(Pointee(Property(&RtpPacketToSend::padding_size, Gt(0u))))));
}
TEST_P(RtpSenderTest, GeneratePaddingCreatesPurePaddingWithoutRtx) {
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
ASSERT_EQ(0,
rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
const size_t kPayloadPacketSize = 1234;
// Send a dummy video packet so it ends up in the packet history. Since we
// are not using RTX, it should never be used as padding.
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Payload padding not available without RTX, only generate plain padding on
// the media SSRC.
// Number of padding packets is the requested padding size divided by max
// padding packet size, rounded up. Pure padding packets are always of the