Google Git
Sign in
webrtc / src / 4fbc3fc59e75615a17722cb93ae0d18caff5ee2e / . / modules / rtp_rtcp / source / rtp_sender_unittest.cc
blob: 0e3ecd4506300cb9f1ff6f29d67b8703e749037b [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 "modules/rtp_rtcp/source/video_fec_generator.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/logging.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/task_utils/to_queued_task.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"
#include "test/time_controller/simulated_time_controller.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 uint64_t kStartTime = 123456789;
const size_t kMaxPaddingSize = 224u;
const uint8_t kPayloadData[] = {47, 11, 32, 93, 89};
const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
const size_t kMaxPaddingLength = 224; // Value taken from rtp_sender.cc.
const uint32_t kTimestampTicksPerMs = 90; // 90kHz clock.
using ::testing::_;
using ::testing::AllOf;
using ::testing::AtLeast;
using ::testing::Contains;
using ::testing::Each;
using ::testing::ElementsAre;
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;
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_METHOD(void,
EnqueuePackets,
(std::vector<std::unique_ptr<RtpPacketToSend>>),
(override));
};
class MockSendSideDelayObserver : public SendSideDelayObserver {
public:
MOCK_METHOD(void,
SendSideDelayUpdated,
(int, int, uint64_t, uint32_t),
(override));
};
class MockSendPacketObserver : public SendPacketObserver {
public:
MOCK_METHOD(void, OnSendPacket, (uint16_t, int64_t, uint32_t), (override));
};
class MockTransportFeedbackObserver : public TransportFeedbackObserver {
public:
MOCK_METHOD(void, OnAddPacket, (const RtpPacketSendInfo&), (override));
MOCK_METHOD(void,
OnTransportFeedback,
(const rtcp::TransportFeedback&),
(override));
};
class TaskQueuePacketSender : public RtpPacketSender {
public:
TaskQueuePacketSender(TimeController* time_controller,
std::unique_ptr<RtpPacketSender> packet_sender)
: time_controller_(time_controller),
packet_sender_(std::move(packet_sender)),
queue_(time_controller_->CreateTaskQueueFactory()->CreateTaskQueue(
"PacerQueue",
TaskQueueFactory::Priority::NORMAL)) {}
void EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) override {
queue_->PostTask(ToQueuedTask([sender = packet_sender_.get(),
packets_ = std::move(packets)]() mutable {
sender->EnqueuePackets(std::move(packets_));
}));
// Trigger task we just enqueued to be executed by updating the simulated
// time controller.
time_controller_->AdvanceTime(TimeDelta::Zero());
}
TaskQueueBase* task_queue() const { return queue_.get(); }
TimeController* const time_controller_;
std::unique_ptr<RtpPacketSender> packet_sender_;
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> queue_;
};
// Mimics ModuleRtpRtcp::RtpSenderContext.
// TODO(sprang): Split up unit tests and test these components individually
// wherever possible.
struct RtpSenderContext : public SequenceNumberAssigner {
RtpSenderContext(const RtpRtcpInterface::Configuration& config,
TimeController* time_controller)
: time_controller_(time_controller),
packet_history_(config.clock, config.enable_rtx_padding_prioritization),
packet_sender_(config, &packet_history_),
pacer_(time_controller,
std::make_unique<RtpSenderEgress::NonPacedPacketSender>(
&packet_sender_,
this)),
packet_generator_(config,
&packet_history_,
config.paced_sender ? config.paced_sender : &pacer_) {
}
void AssignSequenceNumber(RtpPacketToSend* packet) override {
packet_generator_.AssignSequenceNumber(packet);
}
// Inject packet straight into RtpSenderEgress without passing through the
// pacer, but while still running on the pacer task queue.
void InjectPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& packet_info) {
pacer_.task_queue()->PostTask(
ToQueuedTask([sender_ = &packet_sender_, packet_ = std::move(packet),
packet_info]() mutable {
sender_->SendPacket(packet_.get(), packet_info);
}));
time_controller_->AdvanceTime(TimeDelta::Zero());
}
TimeController* time_controller_;
RtpPacketHistory packet_history_;
RtpSenderEgress packet_sender_;
TaskQueuePacketSender pacer_;
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()
: time_controller_(Timestamp::Millis(kStartTime)),
clock_(time_controller_.GetClock()),
retransmission_rate_limiter_(clock_, 1000),
flexfec_sender_(0,
kFlexFecSsrc,
kSsrc,
"",
std::vector<RtpExtension>(),
std::vector<RtpExtensionSize>(),
nullptr,
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) {
SetUpRtpSender(pacer, populate_network2, always_send_mid_and_rid,
&flexfec_sender_);
}
void SetUpRtpSender(bool pacer,
bool populate_network2,
bool always_send_mid_and_rid,
VideoFecGenerator* fec_generator) {
RtpRtcpInterface::Configuration config;
config.clock = clock_;
config.outgoing_transport = &transport_;
config.local_media_ssrc = kSsrc;
config.rtx_send_ssrc = kRtxSsrc;
config.fec_generator = fec_generator;
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.always_send_mid_and_rid = always_send_mid_and_rid;
config.field_trials = &field_trials_;
rtp_sender_context_ =
std::make_unique<RtpSenderContext>(config, &time_controller_);
rtp_sender()->SetSequenceNumber(kSeqNum);
rtp_sender()->SetTimestampOffset(0);
}
GlobalSimulatedTimeController time_controller_;
Clock* const 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_;
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 = clock_->TimeInMilliseconds();
// Use maximum allowed size to catch corner cases when packet is dropped
// because of lack of capacity for the media packet, or for an rtx packet
// containing the media packet.
return SendPacket(kCaptureTimeMs,
/*payload_length=*/rtp_sender()->MaxRtpPacketSize() -
rtp_sender()->ExpectedPerPacketOverhead());
}
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_sender_context_->InjectPacket(std::move(packet), 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(RtpMid::kUri, 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(RtpStreamId::kUri,
kRidExtensionId);
rtp_sender()->RegisterRtpHeaderExtension(RepairedRtpStreamId::kUri,
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_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransmissionOffset::kUri, kTransmissionTimeOffsetExtensionId));
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(AudioLevel::kUri,
kAudioLevelExtensionId));
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
VideoOrientation::kUri, 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(RtpSenderTest, PaddingAlwaysAllowedOnAudio) {
MockTransport transport;
RtpRtcpInterface::Configuration config;
config.audio = true;
config.clock = 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, &time_controller_);
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(RtpSenderTest, SendToNetworkForwardsPacketsToPacer) {
auto packet = BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, 0);
int64_t now_ms = clock_->TimeInMilliseconds();
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum)),
Pointee(Property(&RtpPacketToSend::capture_time_ms, now_ms))))));
EXPECT_TRUE(
rtp_sender()->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
}
TEST_P(RtpSenderTest, ReSendPacketForwardsPacketsToPacer) {
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
int64_t now_ms = clock_->TimeInMilliseconds();
auto packet = BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, now_ms);
uint16_t seq_no = packet->SequenceNumber();
packet->set_allow_retransmission(true);
rtp_sender_context_->packet_history_.PutRtpPacket(std::move(packet), now_ms);
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum)),
Pointee(Property(&RtpPacketToSend::capture_time_ms, now_ms)),
Pointee(Property(&RtpPacketToSend::packet_type,
RtpPacketMediaType::kRetransmission))))));
EXPECT_TRUE(rtp_sender()->ReSendPacket(seq_no));
}
// This test sends 1 regular video packet, then 4 padding packets, and then
// 1 more regular packet.
TEST_P(RtpSenderTest, SendPadding) {
constexpr int kNumPaddingPackets = 4;
EXPECT_CALL(mock_paced_sender_, EnqueuePackets);
std::unique_ptr<RtpPacketToSend> media_packet =
SendPacket(/*capture_time_ms=*/clock_->TimeInMilliseconds(),
/*payload_size=*/100);
// Wait 50 ms before generating each padding packet.
for (int i = 0; i < kNumPaddingPackets; ++i) {
time_controller_.AdvanceTime(TimeDelta::Millis(50));
const size_t kPaddingTargetBytes = 100; // Request 100 bytes of padding.
// Padding should be sent on the media ssrc, with a continous sequence
// number range. Size will be forced to full pack size and the timestamp
// shall be that of the last media packet.
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber,
media_packet->SequenceNumber() + i + 1)),
Pointee(Property(&RtpPacketToSend::padding_size,
kMaxPaddingLength)),
Pointee(Property(&RtpPacketToSend::Timestamp,
media_packet->Timestamp()))))));
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
rtp_sender()->GeneratePadding(kPaddingTargetBytes,
/*media_has_been_sent=*/true);
ASSERT_THAT(padding_packets, SizeIs(1));
rtp_sender()->SendToNetwork(std::move(padding_packets[0]));
}
// Send a regular video packet again.
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(
&RtpPacketToSend::SequenceNumber,
media_packet->SequenceNumber() + kNumPaddingPackets + 1)),
Pointee(Property(&RtpPacketToSend::Timestamp,
Gt(media_packet->Timestamp())))))));
std::unique_ptr<RtpPacketToSend> next_media_packet =
SendPacket(/*capture_time_ms=*/clock_->TimeInMilliseconds(),
/*payload_size=*/100);
}
TEST_P(RtpSenderTest, NoPaddingAsFirstPacketWithoutBweExtensions) {
EXPECT_THAT(rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
/*media_has_been_sent=*/false),
IsEmpty());
// Don't send padding before media even with RTX.
EnableRtx();
EXPECT_THAT(rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
/*media_has_been_sent=*/false),
IsEmpty());
}
TEST_P(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithTransportCc) {
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
// Padding can't be sent as first packet on media SSRC since we don't know
// what payload type to assign.
EXPECT_THAT(rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
/*media_has_been_sent=*/false),
IsEmpty());
// With transportcc padding can be sent as first packet on the RTX SSRC.
EnableRtx();
EXPECT_THAT(rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
/*media_has_been_sent=*/false),
Not(IsEmpty()));
}
TEST_P(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithAbsSendTime) {
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
// Padding can't be sent as first packet on media SSRC since we don't know
// what payload type to assign.
EXPECT_THAT(rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
/*media_has_been_sent=*/false),
IsEmpty());
// With abs send time, padding can be sent as first packet on the RTX SSRC.
EnableRtx();
EXPECT_THAT(rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
/*media_has_been_sent=*/false),
Not(IsEmpty()));
}
TEST_P(RtpSenderTest, UpdatesTimestampsOnPlainRtxPadding) {
EnableRtx();
// Timestamps as set based on capture time in RtpSenderTest.
const int64_t start_time = clock_->TimeInMilliseconds();
const uint32_t start_timestamp = start_time * kTimestampTicksPerMs;
// Start by sending one media packet.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::padding_size, 0u)),
Pointee(Property(&RtpPacketToSend::Timestamp, start_timestamp)),
Pointee(Property(&RtpPacketToSend::capture_time_ms, start_time))))));
std::unique_ptr<RtpPacketToSend> media_packet =
SendPacket(start_time, /*payload_size=*/600);
// Advance time before sending padding.
const TimeDelta kTimeDiff = TimeDelta::Millis(17);
time_controller_.AdvanceTime(kTimeDiff);
// Timestamps on padding should be offset from the sent media.
EXPECT_THAT(
rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
/*media_has_been_sent=*/true),
Each(AllOf(
Pointee(Property(&RtpPacketToSend::padding_size, kMaxPaddingLength)),
Pointee(Property(
&RtpPacketToSend::Timestamp,
start_timestamp + (kTimestampTicksPerMs * kTimeDiff.ms()))),
Pointee(Property(&RtpPacketToSend::capture_time_ms,
start_time + kTimeDiff.ms())))));
}
TEST_P(RtpSenderTest, KeepsTimestampsOnPayloadPadding) {
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
EnableRtx();
// Timestamps as set based on capture time in RtpSenderTest.
const int64_t start_time = clock_->TimeInMilliseconds();
const uint32_t start_timestamp = start_time * kTimestampTicksPerMs;
const size_t kPayloadSize = 600;
const size_t kRtxHeaderSize = 2;
// Start by sending one media packet and putting in the packet history.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(Contains(AllOf(
Pointee(Property(&RtpPacketToSend::padding_size, 0u)),
Pointee(Property(&RtpPacketToSend::Timestamp, start_timestamp)),
Pointee(Property(&RtpPacketToSend::capture_time_ms, start_time))))));
std::unique_ptr<RtpPacketToSend> media_packet =
SendPacket(start_time, kPayloadSize);
rtp_sender_context_->packet_history_.PutRtpPacket(std::move(media_packet),
start_time);
// Advance time before sending padding.
const TimeDelta kTimeDiff = TimeDelta::Millis(17);
time_controller_.AdvanceTime(kTimeDiff);
// Timestamps on payload padding should be set to original.
EXPECT_THAT(
rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
/*media_has_been_sent=*/true),
Each(AllOf(
Pointee(Property(&RtpPacketToSend::padding_size, 0u)),
Pointee(Property(&RtpPacketToSend::payload_size,
kPayloadSize + kRtxHeaderSize)),
Pointee(Property(&RtpPacketToSend::Timestamp, start_timestamp)),
Pointee(Property(&RtpPacketToSend::capture_time_ms, start_time)))));
}
// 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";
const char kNoRid[] = "";
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(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);
time_controller_.AdvanceTime(TimeDelta::Millis(1));
SendPacket(clock_->TimeInMilliseconds(), kPacketSize);
}
EXPECT_EQ(kNumPackets, transport_.packets_sent());
time_controller_.AdvanceTime(TimeDelta::Millis(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.
time_controller_.AdvanceTime(TimeDelta::Millis(5));
// Resending should not work, bandwidth exceeded.
rtp_sender()->OnReceivedNack(sequence_numbers, 0);
EXPECT_EQ(kNumPackets * 2, transport_.packets_sent());
}
TEST_P(RtpSenderTest, UpdatingCsrcsUpdatedOverhead) {
RtpRtcpInterface::Configuration config;
config.clock = 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, &time_controller_);
// 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) {
RtpRtcpInterface::Configuration config;
config.clock = 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, &time_controller_);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
rtp_sender()->RegisterRtpHeaderExtension(TransmissionOffset::kUri,
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) {
RtpRtcpInterface::Configuration config;
config.clock = 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, &time_controller_);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
rtp_sender()->RegisterRtpHeaderExtension(RtpMid::kUri, kMidExtensionId);
rtp_sender()->RegisterRtpHeaderExtension(RtpStreamId::kUri, 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) {
RtpRtcpInterface::Configuration config;
config.clock = 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, &time_controller_);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
rtp_sender()->RegisterRtpHeaderExtension(InbandComfortNoiseExtension::kUri,
1);
rtp_sender()->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::kUri,
2);
rtp_sender()->RegisterRtpHeaderExtension(VideoOrientation::kUri, 3);
rtp_sender()->RegisterRtpHeaderExtension(PlayoutDelayLimits::kUri, 4);
rtp_sender()->RegisterRtpHeaderExtension(VideoContentTypeExtension::kUri, 5);
rtp_sender()->RegisterRtpHeaderExtension(VideoTimingExtension::kUri, 6);
rtp_sender()->RegisterRtpHeaderExtension(RepairedRtpStreamId::kUri, 7);
rtp_sender()->RegisterRtpHeaderExtension(ColorSpaceExtension::kUri, 8);
// Still only 12B counted since can't count on above being sent.
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(), 12u);
}
TEST_P(RtpSenderTest, SendPacketHandlesRetransmissionHistory) {
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
// Ignore calls to EnqueuePackets() for this test.
EXPECT_CALL(mock_paced_sender_, EnqueuePackets).WillRepeatedly(Return());
// Build a media packet and put in the packet history.
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, clock_->TimeInMilliseconds());
const uint16_t media_sequence_number = packet->SequenceNumber();
packet->set_allow_retransmission(true);
rtp_sender_context_->packet_history_.PutRtpPacket(
std::move(packet), clock_->TimeInMilliseconds());
// Simulate successful retransmission request.
time_controller_.AdvanceTime(TimeDelta::Millis(30));
EXPECT_THAT(rtp_sender()->ReSendPacket(media_sequence_number), Gt(0));
// Packet already pending, retransmission not allowed.
time_controller_.AdvanceTime(TimeDelta::Millis(30));
EXPECT_THAT(rtp_sender()->ReSendPacket(media_sequence_number), Eq(0));
// Simulate packet exiting pacer, mark as not longer pending.
rtp_sender_context_->packet_history_.MarkPacketAsSent(media_sequence_number);
// Retransmissions allowed again.
time_controller_.AdvanceTime(TimeDelta::Millis(30));
EXPECT_THAT(rtp_sender()->ReSendPacket(media_sequence_number), Gt(0));
}
TEST_P(RtpSenderTest, MarksRetransmittedPackets) {
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
// Build a media packet and put in the packet history.
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, clock_->TimeInMilliseconds());
const uint16_t media_sequence_number = packet->SequenceNumber();
packet->set_allow_retransmission(true);
rtp_sender_context_->packet_history_.PutRtpPacket(
std::move(packet), clock_->TimeInMilliseconds());
// Expect a retransmission packet marked with which packet it is a
// retransmit of.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(AllOf(
Pointee(Property(&RtpPacketToSend::packet_type,
RtpPacketMediaType::kRetransmission)),
Pointee(Property(&RtpPacketToSend::retransmitted_sequence_number,
Eq(media_sequence_number)))))));
EXPECT_THAT(rtp_sender()->ReSendPacket(media_sequence_number), Gt(0));
}
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_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransmissionOffset::kUri, kTransmissionTimeOffsetExtensionId));
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
// Send a payload packet first, to enable padding and populate the packet
// history.
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, 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_sender_context_->InjectPacket(std::move(packet), 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_sender_context_->InjectPacket(std::move(plain_padding),
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_sender_context_->InjectPacket(std::move(payload_padding),
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_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
const size_t kPayloadPacketSize = kMinPaddingSize;
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, 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_sender_context_->InjectPacket(std::move(packet), 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_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, 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, 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_sender_context_->InjectPacket(std::move(packet), 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_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransmissionOffset::kUri, kTransmissionTimeOffsetExtensionId));
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, 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, 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_sender_context_->InjectPacket(std::move(packet), 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
// maximum size.
const size_t kPaddingBytesRequested = kPayloadPacketSize + kRtxHeaderSize;
const size_t kExpectedNumPaddingPackets =
(kPaddingBytesRequested + kMaxPaddingSize - 1) / kMaxPaddingSize;
size_t padding_bytes_generated = 0;
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
rtp_sender()->GeneratePadding(kPaddingBytesRequested, true);
EXPECT_EQ(padding_packets.size(), kExpectedNumPaddingPackets);
for (auto& packet : padding_packets) {
EXPECT_EQ(packet->packet_type(), RtpPacketMediaType::kPadding);
EXPECT_EQ(packet->Ssrc(), kSsrc);
EXPECT_EQ(packet->payload_size(), 0u);
EXPECT_GT(packet->padding_size(), 0u);
padding_bytes_generated += packet->padding_size();
EXPECT_TRUE(packet->HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(packet->HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(packet->HasExtension<TransmissionOffset>());
// Verify all header extensions are received.
rtp_sender_context_->InjectPacket(std::move(packet), PacedPacketInfo());
webrtc::RTPHeader rtp_header;
transport_.last_sent_packet().GetHeader(&rtp_header);
EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
}
EXPECT_EQ(padding_bytes_generated,
kExpectedNumPaddingPackets * kMaxPaddingSize);
}
TEST_P(RtpSenderTest, SupportsPadding) {
bool kSendingMediaStats[] = {true, false};
bool kEnableRedundantPayloads[] = {true, false};
absl::string_view kBweExtensionUris[] = {
TransportSequenceNumber::kUri, TransportSequenceNumberV2::kUri,
AbsoluteSendTime::kUri, TransmissionOffset::kUri};
const int kExtensionsId = 7;
for (bool sending_media : kSendingMediaStats) {
rtp_sender()->SetSendingMediaStatus(sending_media);
for (bool redundant_payloads : kEnableRedundantPayloads) {
int rtx_mode = kRtxRetransmitted;
if (redundant_payloads) {
rtx_mode |= kRtxRedundantPayloads;
}
rtp_sender()->SetRtxStatus(rtx_mode);
for (auto extension_uri : kBweExtensionUris) {
EXPECT_FALSE(rtp_sender()->SupportsPadding());
rtp_sender()->RegisterRtpHeaderExtension(extension_uri, kExtensionsId);
if (!sending_media) {
EXPECT_FALSE(rtp_sender()->SupportsPadding());
} else {
EXPECT_TRUE(rtp_sender()->SupportsPadding());
if (redundant_payloads) {
EXPECT_TRUE(rtp_sender()->SupportsRtxPayloadPadding());
} else {
EXPECT_FALSE(rtp_sender()->SupportsRtxPayloadPadding());
}
}
rtp_sender()->DeregisterRtpHeaderExtension(extension_uri);
EXPECT_FALSE(rtp_sender()->SupportsPadding());
}
}
}
}
TEST_P(RtpSenderTest, SetsCaptureTimeOnRtxRetransmissions) {
EnableRtx();
// Put a packet in the packet history, with current time as capture time.
const int64_t start_time_ms = clock_->TimeInMilliseconds();
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, kMarkerBit, start_time_ms,
/*capture_time_ms=*/start_time_ms);
packet->set_allow_retransmission(true);
rtp_sender_context_->packet_history_.PutRtpPacket(std::move(packet),
start_time_ms);
// Advance time, request an RTX retransmission. Capture timestamp should be
// preserved.
time_controller_.AdvanceTime(TimeDelta::Millis(10));
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(Property(
&RtpPacketToSend::capture_time_ms, start_time_ms)))));
EXPECT_GT(rtp_sender()->ReSendPacket(kSeqNum), 0);
}
TEST_P(RtpSenderTestWithoutPacer, ClearHistoryOnSequenceNumberCange) {
EnableRtx();
// Put a packet in the packet history.
const int64_t now_ms = clock_->TimeInMilliseconds();
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, kMarkerBit, now_ms, now_ms);
packet->set_allow_retransmission(true);
rtp_sender_context_->packet_history_.PutRtpPacket(std::move(packet), now_ms);
EXPECT_TRUE(rtp_sender_context_->packet_history_.GetPacketState(kSeqNum));
// Update the sequence number of the RTP module, verify packet has been
// removed.
rtp_sender()->SetSequenceNumber(rtp_sender()->SequenceNumber() - 1);
EXPECT_FALSE(rtp_sender_context_->packet_history_.GetPacketState(kSeqNum));
}
TEST_P(RtpSenderTest, IgnoresNackAfterDisablingMedia) {
const int64_t kRtt = 10;
EnableRtx();
rtp_sender_context_->packet_history_.SetRtt(kRtt);
// Put a packet in the history.
const int64_t start_time_ms = clock_->TimeInMilliseconds();
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, kMarkerBit, start_time_ms,
/*capture_time_ms=*/start_time_ms);
packet->set_allow_retransmission(true);
rtp_sender_context_->packet_history_.PutRtpPacket(std::move(packet),
start_time_ms);
// Disable media sending and try to retransmit the packet, it should fail.
rtp_sender()->SetSendingMediaStatus(false);
time_controller_.AdvanceTime(TimeDelta::Millis(kRtt));
EXPECT_LT(rtp_sender()->ReSendPacket(kSeqNum), 0);
}
TEST_P(RtpSenderTest, DoesntFecProtectRetransmissions) {
// Set up retranmission without RTX, so that a plain copy of the old packet is
// re-sent instead.
const int64_t kRtt = 10;
rtp_sender()->SetSendingMediaStatus(true);
rtp_sender()->SetRtxStatus(kRtxOff);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
rtp_sender_context_->packet_history_.SetRtt(kRtt);
// Put a fec protected packet in the history.
const int64_t start_time_ms = clock_->TimeInMilliseconds();
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, kMarkerBit, start_time_ms,
/*capture_time_ms=*/start_time_ms);
packet->set_allow_retransmission(true);
packet->set_fec_protect_packet(true);
rtp_sender_context_->packet_history_.PutRtpPacket(std::move(packet),
start_time_ms);
// Re-send packet, the retransmitted packet should not have the FEC protection
// flag set.
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(
Property(&RtpPacketToSend::fec_protect_packet, false)))));
time_controller_.AdvanceTime(TimeDelta::Millis(kRtt));
EXPECT_GT(rtp_sender()->ReSendPacket(kSeqNum), 0);
}
TEST_P(RtpSenderTest, MarksPacketsWithKeyframeStatus) {
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = 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
{
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(Each(
Pointee(Property(&RtpPacketToSend::is_key_frame, true)))))
.Times(AtLeast(1));
RTPVideoHeader video_header;
video_header.frame_type = VideoFrameType::kVideoFrameKey;
int64_t capture_time_ms = clock_->TimeInMilliseconds();
EXPECT_TRUE(rtp_sender_video.SendVideo(
kPayloadType, kCodecType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, video_header, kDefaultExpectedRetransmissionTimeMs));
time_controller_.AdvanceTime(TimeDelta::Millis(33));
}
{
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(Each(
Pointee(Property(&RtpPacketToSend::is_key_frame, false)))))
.Times(AtLeast(1));
RTPVideoHeader video_header;
video_header.frame_type = VideoFrameType::kVideoFrameDelta;
int64_t capture_time_ms = clock_->TimeInMilliseconds();
EXPECT_TRUE(rtp_sender_video.SendVideo(
kPayloadType, kCodecType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, video_header, kDefaultExpectedRetransmissionTimeMs));
time_controller_.AdvanceTime(TimeDelta::Millis(33));
}
}
INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
RtpSenderTest,
::testing::Values(TestConfig{false},
TestConfig{true}));
INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
RtpSenderTestWithoutPacer,
::testing::Values(TestConfig{false},
TestConfig{true}));
} // namespace webrtc