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webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
Danil Chapovalov c54c85fe8f Attach Mid/Rid RTP header extension to pure padding packets 
same as they attached to other packets.
Otherwise there is risk that ssrc will be acked after few initial pure padding packets are sent, before remote endpoint seen any mid or rid attached.

Bug: b/361257385
Change-Id: I695b379221debe2518ad33d13d65620877f0b2a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360660
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42851}
2024-08-26 16:00:16 +00:00

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/*
* 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 "absl/strings/string_view.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/rtc_event_log/rtc_event.h"
#include "api/rtp_packet_sender.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.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_rtcp_defines.h"
#include "modules/rtp_rtcp/source/packet_sequencer.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_video.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 "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_transport.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 uint8_t kPayloadData[] = {47, 11, 32, 93, 89};
constexpr TimeDelta kDefaultExpectedRetransmissionTime = TimeDelta::Millis(125);
constexpr Frequency kRtpClockRate = Frequency::Hertz(90'000);
constexpr absl::string_view kMid = "mid";
constexpr absl::string_view kRid = "f";
constexpr bool kMarkerBit = true;
using ::testing::_;
using ::testing::AllOf;
using ::testing::AtLeast;
using ::testing::Contains;
using ::testing::Each;
using ::testing::ElementsAre;
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;
class MockRtpPacketPacer : public RtpPacketSender {
public:
MockRtpPacketPacer() {}
virtual ~MockRtpPacketPacer() {}
MOCK_METHOD(void,
EnqueuePackets,
(std::vector<std::unique_ptr<RtpPacketToSend>>),
(override));
MOCK_METHOD(void, RemovePacketsForSsrc, (uint32_t), (override));
};
uint32_t ToRtpTimestamp(Timestamp time) {
return static_cast<int64_t>((time - Timestamp::Zero()) * kRtpClockRate) &
0xFFFF'FFFF;
}
} // namespace
class RtpSenderTest : public ::testing::Test {
protected:
RtpSenderTest()
: time_controller_(Timestamp::Millis(kStartTime)),
env_(CreateEnvironment(time_controller_.GetClock())),
retransmission_rate_limiter_(&env_.clock(), 1000),
flexfec_sender_(env_,
0,
kFlexFecSsrc,
kSsrc,
"",
std::vector<RtpExtension>(),
std::vector<RtpExtensionSize>(),
nullptr) {}
void SetUp() override { SetUpRtpSender(true, false, nullptr); }
void SetUpRtpSender(bool populate_network2,
bool always_send_mid_and_rid,
VideoFecGenerator* fec_generator) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
config.fec_generator = fec_generator;
config.populate_network2_timestamp = populate_network2;
config.always_send_mid_and_rid = always_send_mid_and_rid;
CreateSender(config);
}
RtpRtcpInterface::Configuration GetDefaultConfig() {
RtpRtcpInterface::Configuration config;
config.clock = &env_.clock();
config.local_media_ssrc = kSsrc;
config.rtx_send_ssrc = kRtxSsrc;
config.event_log = &mock_rtc_event_log_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.paced_sender = &mock_paced_sender_;
config.field_trials = &env_.field_trials();
// Configure rid unconditionally, it has effect only if
// corresponding header extension is enabled.
config.rid = std::string(kRid);
return config;
}
void CreateSender(const RtpRtcpInterface::Configuration& config) {
packet_history_ = std::make_unique<RtpPacketHistory>(
config.clock, RtpPacketHistory::PaddingMode::kRecentLargePacket);
sequencer_.emplace(kSsrc, kRtxSsrc,
/*require_marker_before_media_padding=*/!config.audio,
&env_.clock());
rtp_sender_ = std::make_unique<RTPSender>(config, packet_history_.get(),
config.paced_sender);
sequencer_->set_media_sequence_number(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
}
GlobalSimulatedTimeController time_controller_;
const Environment env_;
NiceMock<MockRtcEventLog> mock_rtc_event_log_;
MockRtpPacketPacer mock_paced_sender_;
RateLimiter retransmission_rate_limiter_;
FlexfecSender flexfec_sender_;
absl::optional<PacketSequencer> sequencer_;
std::unique_ptr<RtpPacketHistory> packet_history_;
std::unique_ptr<RTPSender> rtp_sender_;
std::unique_ptr<RtpPacketToSend> BuildRtpPacket(int payload_type,
bool marker_bit,
uint32_t rtp_timestamp,
Timestamp capture_time) {
auto packet = rtp_sender_->AllocatePacket();
packet->SetPayloadType(payload_type);
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet->SetMarker(marker_bit);
packet->SetTimestamp(rtp_timestamp);
packet->set_capture_time(capture_time);
return packet;
}
std::unique_ptr<RtpPacketToSend> SendPacket(Timestamp capture_time,
int payload_length) {
uint32_t rtp_timestamp = ToRtpTimestamp(capture_time);
auto packet =
BuildRtpPacket(kPayload, kMarkerBit, rtp_timestamp, capture_time);
packet->AllocatePayload(payload_length);
packet->set_allow_retransmission(true);
// Packet should be stored in a send bucket.
std::vector<std::unique_ptr<RtpPacketToSend>> packets(1);
packets[0] = std::make_unique<RtpPacketToSend>(*packet);
rtp_sender_->EnqueuePackets(std::move(packets));
return packet;
}
std::unique_ptr<RtpPacketToSend> SendGenericPacket() {
// 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(/*capture_time=*/env_.clock().CurrentTime(),
/*payload_length=*/rtp_sender_->MaxRtpPacketSize() -
rtp_sender_->ExpectedPerPacketOverhead());
}
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes) {
return rtp_sender_->GeneratePadding(
target_size_bytes, /*media_has_been_sent=*/true,
sequencer_->CanSendPaddingOnMediaSsrc());
}
std::vector<std::unique_ptr<RtpPacketToSend>> Sequence(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
for (auto& packet : packets) {
sequencer_->Sequence(*packet);
}
return packets;
}
size_t GenerateAndSendPadding(size_t target_size_bytes) {
size_t generated_bytes = 0;
std::vector<std::unique_ptr<RtpPacketToSend>> packets;
for (auto& packet : GeneratePadding(target_size_bytes)) {
generated_bytes += packet->payload_size() + packet->padding_size();
packets.push_back(std::move(packet));
}
rtp_sender_->EnqueuePackets(std::move(packets));
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;
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(absl::string_view mid) {
rtp_sender_->RegisterRtpHeaderExtension(RtpMid::Uri(), 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() {
rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
kRidExtensionId);
rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(),
kRepairedRidExtensionId);
}
};
TEST_F(RtpSenderTest, AllocatePacketSetCsrcs) {
// Configure rtp_sender with csrc.
uint32_t csrcs[] = {0x23456789};
auto packet = rtp_sender_->AllocatePacket(csrcs);
ASSERT_TRUE(packet);
EXPECT_EQ(rtp_sender_->SSRC(), packet->Ssrc());
EXPECT_THAT(packet->Csrcs(), ElementsAreArray(csrcs));
}
TEST_F(RtpSenderTest, AllocatePacketReserveExtensions) {
// Configure rtp_sender with extensions.
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransmissionOffset::Uri(), kTransmissionTimeOffsetExtensionId));
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
AudioLevelExtension::Uri(), kAudioLevelExtensionId));
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
VideoOrientation::Uri(), 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<AudioLevelExtension>());
EXPECT_FALSE(packet->HasExtension<VideoOrientation>());
}
TEST_F(RtpSenderTest, PaddingAlwaysAllowedOnAudio) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
config.audio = true;
CreateSender(config);
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);
sequencer_->Sequence(*audio_packet);
const size_t kPaddingSize = 59;
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(AllOf(
Pointee(Property(&RtpPacketToSend::packet_type,
RtpPacketMediaType::kPadding)),
Pointee(Property(&RtpPacketToSend::padding_size, kPaddingSize))))));
EXPECT_EQ(kPaddingSize, GenerateAndSendPadding(kPaddingSize));
// Requested padding size is too small, will send a larger one.
const size_t kMinPaddingSize = 50;
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(
AllOf(Pointee(Property(&RtpPacketToSend::packet_type,
RtpPacketMediaType::kPadding)),
Pointee(Property(&RtpPacketToSend::padding_size,
kMinPaddingSize))))));
EXPECT_EQ(kMinPaddingSize, GenerateAndSendPadding(kMinPaddingSize - 5));
}
TEST_F(RtpSenderTest, SendToNetworkForwardsPacketsToPacer) {
std::vector<std::unique_ptr<RtpPacketToSend>> packets(1);
packets[0] =
BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, Timestamp::Zero());
Timestamp now = env_.clock().CurrentTime();
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::capture_time, now))))));
rtp_sender_->EnqueuePackets(std::move(packets));
}
TEST_F(RtpSenderTest, ReSendPacketForwardsPacketsToPacer) {
packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
Timestamp now = env_.clock().CurrentTime();
auto packet = BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, now);
packet->SetSequenceNumber(kSeqNum);
packet->set_allow_retransmission(true);
packet_history_->PutRtpPacket(std::move(packet), now);
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum)),
Pointee(Property(&RtpPacketToSend::capture_time, now)),
Pointee(Property(&RtpPacketToSend::packet_type,
RtpPacketMediaType::kRetransmission))))));
EXPECT_TRUE(rtp_sender_->ReSendPacket(kSeqNum));
}
// This test sends 1 regular video packet, then 4 padding packets, and then
// 1 more regular packet.
TEST_F(RtpSenderTest, SendPadding) {
constexpr int kNumPaddingPackets = 4;
EXPECT_CALL(mock_paced_sender_, EnqueuePackets);
std::unique_ptr<RtpPacketToSend> media_packet =
SendPacket(/*capture_time=*/env_.clock().CurrentTime(),
/*payload_size=*/100);
sequencer_->Sequence(*media_packet);
// 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(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::Ssrc, kSsrc),
Property(&RtpPacketToSend::padding_size, kMaxPaddingLength),
Property(&RtpPacketToSend::SequenceNumber,
media_packet->SequenceNumber() + i + 1),
Property(&RtpPacketToSend::Timestamp,
media_packet->Timestamp()))))));
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
Sequence(GeneratePadding(kPaddingTargetBytes));
ASSERT_THAT(padding_packets, SizeIs(1));
rtp_sender_->EnqueuePackets(std::move(padding_packets));
}
// Send a regular video packet again.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(Property(
&RtpPacketToSend::Timestamp, Gt(media_packet->Timestamp()))))));
std::unique_ptr<RtpPacketToSend> next_media_packet =
SendPacket(/*capture_time=*/env_.clock().CurrentTime(),
/*payload_size=*/100);
}
TEST_F(RtpSenderTest, NoPaddingAsFirstPacketWithoutBweExtensions) {
EXPECT_THAT(rtp_sender_->GeneratePadding(
/*target_size_bytes=*/100,
/*media_has_been_sent=*/false,
/*can_send_padding_on_media_ssrc=*/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,
/*can_send_padding_on_media_ssrc=*/false),
IsEmpty());
}
TEST_F(RtpSenderTest, RequiresRtxSsrcToEnableRtx) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
config.rtx_send_ssrc = absl::nullopt;
RTPSender rtp_sender(config, packet_history_.get(), config.paced_sender);
rtp_sender.SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender.SetRtxStatus(kRtxRetransmitted);
EXPECT_EQ(rtp_sender.RtxStatus(), kRtxOff);
}
TEST_F(RtpSenderTest, RequiresRtxPayloadTypesToEnableRtx) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
config.rtx_send_ssrc = kRtxSsrc;
RTPSender rtp_sender(config, packet_history_.get(), config.paced_sender);
rtp_sender.SetRtxStatus(kRtxRetransmitted);
EXPECT_EQ(rtp_sender.RtxStatus(), kRtxOff);
}
TEST_F(RtpSenderTest, CanEnableRtxWhenRtxSsrcAndPayloadTypeAreConfigured) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
config.rtx_send_ssrc = kRtxSsrc;
RTPSender rtp_sender(config, packet_history_.get(), config.paced_sender);
rtp_sender.SetRtxPayloadType(kRtxPayload, kPayload);
ASSERT_EQ(rtp_sender.RtxStatus(), kRtxOff);
rtp_sender.SetRtxStatus(kRtxRetransmitted);
EXPECT_EQ(rtp_sender.RtxStatus(), kRtxRetransmitted);
}
TEST_F(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithTransportCc) {
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::Uri(), 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,
/*can_send_padding_on_media_ssrc=*/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,
/*can_send_padding_on_media_ssrc=*/false),
Not(IsEmpty()));
}
TEST_F(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithAbsSendTime) {
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
AbsoluteSendTime::Uri(), 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,
/*can_send_padding_on_media_ssrc=*/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,
/*can_send_padding_on_media_ssrc=*/false),
Not(IsEmpty()));
}
TEST_F(RtpSenderTest, UpdatesTimestampsOnPlainRtxPadding) {
EnableRtx();
// Timestamps as set based on capture time in RtpSenderTest.
const Timestamp start_time = env_.clock().CurrentTime();
const uint32_t start_timestamp = ToRtpTimestamp(start_time);
// Start by sending one media packet.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(AllOf(
Pointee(Property(&RtpPacketToSend::padding_size, 0u)),
Pointee(Property(&RtpPacketToSend::Timestamp, start_timestamp)),
Pointee(Property(&RtpPacketToSend::capture_time, start_time))))));
std::unique_ptr<RtpPacketToSend> media_packet =
SendPacket(start_time, /*payload_size=*/600);
sequencer_->Sequence(*media_packet);
// 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(
Sequence(GeneratePadding(/*target_size_bytes=*/100)),
Each(Pointee(AllOf(
Property(&RtpPacketToSend::padding_size, kMaxPaddingLength),
Property(&RtpPacketToSend::Timestamp,
start_timestamp + kRtpClockRate * kTimeDiff),
Property(&RtpPacketToSend::capture_time, start_time + kTimeDiff)))));
}
TEST_F(RtpSenderTest, KeepsTimestampsOnPayloadPadding) {
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
EnableRtx();
// Timestamps as set based on capture time in RtpSenderTest.
const Timestamp start_time = env_.clock().CurrentTime();
const uint32_t start_timestamp = ToRtpTimestamp(start_time);
const size_t kPayloadSize = 200;
const size_t kRtxHeaderSize = 2;
// Start by sending one media packet and putting in the packet history.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(AllOf(
Pointee(Property(&RtpPacketToSend::padding_size, 0u)),
Pointee(Property(&RtpPacketToSend::Timestamp, start_timestamp)),
Pointee(Property(&RtpPacketToSend::capture_time, start_time))))));
std::unique_ptr<RtpPacketToSend> media_packet =
SendPacket(start_time, kPayloadSize);
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(
GeneratePadding(/*target_size_bytes=*/100),
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, start_time)))));
}
// Test that the MID header extension is included on sent packets when
// configured.
TEST_F(RtpSenderTest, MidIncludedOnSentPackets) {
EnableMidSending(kMid);
// Send a couple packets, expect both packets to have the MID set.
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(
Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid)))))
.Times(2);
SendGenericPacket();
SendGenericPacket();
}
TEST_F(RtpSenderTest, RidIncludedOnSentPackets) {
EnableRidSending();
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(Property(
&RtpPacketToSend::GetExtension<RtpStreamId>, kRid)))));
SendGenericPacket();
}
TEST_F(RtpSenderTest, RidIncludedOnRtxSentPackets) {
EnableRtx();
EnableRidSending();
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid),
Property(&RtpPacketToSend::HasExtension<RepairedRtpStreamId>,
false))))))
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
sequencer_->Sequence(*packets[0]);
packet_history_->PutRtpPacket(std::move(packets[0]),
env_.clock().CurrentTime());
});
SendGenericPacket();
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::GetExtension<RepairedRtpStreamId>, kRid),
Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
rtp_sender_->ReSendPacket(kSeqNum);
}
TEST_F(RtpSenderTest, MidAndRidNotIncludedOnSentPacketsAfterAck) {
EnableMidSending(kMid);
EnableRidSending();
// This first packet should include both MID and RID.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid))))));
auto first_built_packet = SendGenericPacket();
rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
// The second packet should include neither since an ack was received.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::HasExtension<RtpMid>, false),
Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
SendGenericPacket();
}
TEST_F(RtpSenderTest, MidAndRidAlwaysIncludedOnSentPacketsWhenConfigured) {
SetUpRtpSender(false, /*always_send_mid_and_rid=*/true, nullptr);
EnableMidSending(kMid);
EnableRidSending();
// Send two media packets: one before and one after the ack.
// Due to the configuration, both sent packets should contain MID and RID.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(
AllOf(Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid))))))
.Times(2);
auto first_built_packet = SendGenericPacket();
rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
SendGenericPacket();
}
// 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_F(RtpSenderTest, MidAndRidIncludedOnFirstRtxPacket) {
EnableRtx();
EnableMidSending(kMid);
EnableRidSending();
// This first packet will include both MID and RID.
EXPECT_CALL(mock_paced_sender_, EnqueuePackets);
auto first_built_packet = SendGenericPacket();
rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
// The second packet will include neither since an ack was received, put
// it in the packet history for retransmission.
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
packet_history_->PutRtpPacket(std::move(packets[0]),
env_.clock().CurrentTime());
});
auto second_built_packet = SendGenericPacket();
// The first RTX packet should include MID and RRID.
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
Property(&RtpPacketToSend::GetExtension<RepairedRtpStreamId>,
kRid))))));
rtp_sender_->ReSendPacket(second_built_packet->SequenceNumber());
}
// 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_F(RtpSenderTest, MidAndRidNotIncludedOnRtxPacketsAfterAck) {
EnableRtx();
EnableMidSending(kMid);
EnableRidSending();
// This first packet will include both MID and RID.
auto first_built_packet = SendGenericPacket();
sequencer_->Sequence(*first_built_packet);
packet_history_->PutRtpPacket(
std::make_unique<RtpPacketToSend>(*first_built_packet),
/*send_time=*/env_.clock().CurrentTime());
rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
// The second packet will include neither since an ack was received.
auto second_built_packet = SendGenericPacket();
sequencer_->Sequence(*second_built_packet);
packet_history_->PutRtpPacket(
std::make_unique<RtpPacketToSend>(*second_built_packet),
/*send_time=*/env_.clock().CurrentTime());
// The first RTX packet will include MID and RRID.
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
rtp_sender_->OnReceivedAckOnRtxSsrc(packets[0]->SequenceNumber());
packet_history_->MarkPacketAsSent(
*packets[0]->retransmitted_sequence_number());
});
rtp_sender_->ReSendPacket(second_built_packet->SequenceNumber());
// The second and third RTX packets should not include MID nor RRID.
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::HasExtension<RtpMid>, false),
Property(&RtpPacketToSend::HasExtension<RepairedRtpStreamId>,
false))))))
.Times(2);
rtp_sender_->ReSendPacket(first_built_packet->SequenceNumber());
rtp_sender_->ReSendPacket(second_built_packet->SequenceNumber());
}
TEST_F(RtpSenderTest, MidAndRidAlwaysIncludedOnRtxPacketsWhenConfigured) {
SetUpRtpSender(false, /*always_send_mid_and_rid=*/true, nullptr);
EnableRtx();
EnableMidSending(kMid);
EnableRidSending();
// Send two media packets: one before and one after the ack.
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(
AllOf(Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid))))))
.Times(2)
.WillRepeatedly(
[&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
packet_history_->PutRtpPacket(std::move(packets[0]),
env_.clock().CurrentTime());
});
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.
// Due to the configuration, all sent packets should contain MID
// and either RID (media) or RRID (RTX).
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
Property(&RtpPacketToSend::GetExtension<RepairedRtpStreamId>,
kRid))))))
.Times(3)
.WillRepeatedly(
[&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
rtp_sender_->OnReceivedAckOnRtxSsrc(packets[0]->SequenceNumber());
packet_history_->MarkPacketAsSent(
*packets[0]->retransmitted_sequence_number());
});
rtp_sender_->ReSendPacket(media_packet2->SequenceNumber());
rtp_sender_->ReSendPacket(media_packet1->SequenceNumber());
rtp_sender_->ReSendPacket(media_packet2->SequenceNumber());
}
// 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_F(RtpSenderTest, MidAndRidNotIncludedOnSentPacketsAfterRtpStateRestored) {
EnableMidSending(kMid);
EnableRidSending();
RtpState state = rtp_sender_->GetRtpState();
EXPECT_FALSE(state.ssrc_has_acked);
state.ssrc_has_acked = true;
rtp_sender_->SetRtpState(state);
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::HasExtension<RtpMid>, false),
Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
SendGenericPacket();
}
// 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_F(RtpSenderTest, MidAndRridNotIncludedOnRtxPacketsAfterRtpStateRestored) {
EnableRtx();
EnableMidSending(kMid);
EnableRidSending();
RtpState rtx_state = rtp_sender_->GetRtxRtpState();
EXPECT_FALSE(rtx_state.ssrc_has_acked);
rtx_state.ssrc_has_acked = true;
rtp_sender_->SetRtxRtpState(rtx_state);
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
packet_history_->PutRtpPacket(std::move(packets[0]),
env_.clock().CurrentTime());
});
auto built_packet = SendGenericPacket();
EXPECT_CALL(
mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::HasExtension<RtpMid>, false),
Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
ASSERT_LT(0, rtp_sender_->ReSendPacket(built_packet->SequenceNumber()));
}
TEST_F(RtpSenderTest, RespectsNackBitrateLimit) {
const int32_t kPacketSize = 1400;
const int32_t kNumPackets = 30;
retransmission_rate_limiter_.SetMaxRate(kPacketSize * kNumPackets * 8);
EnableRtx();
std::vector<uint16_t> sequence_numbers;
for (int32_t i = 0; i < kNumPackets; ++i) {
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, /*marker_bit=*/true, /*rtp_timestamp=*/0,
/*capture_time=*/env_.clock().CurrentTime());
packet->set_allow_retransmission(true);
sequencer_->Sequence(*packet);
sequence_numbers.push_back(packet->SequenceNumber());
packet_history_->PutRtpPacket(std::move(packet),
/*send_time=*/env_.clock().CurrentTime());
time_controller_.AdvanceTime(TimeDelta::Millis(1));
}
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).
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(ElementsAre(Pointee(Property(
&RtpPacketToSend::packet_type,
RtpPacketMediaType::kRetransmission)))))
.Times(kNumPackets)
.WillRepeatedly(
[&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
for (const auto& packet : packets) {
packet_history_->MarkPacketAsSent(
*packet->retransmitted_sequence_number());
}
});
rtp_sender_->OnReceivedNack(sequence_numbers, 0);
// Must be at least 5ms in between retransmission attempts.
time_controller_.AdvanceTime(TimeDelta::Millis(5));
// Resending should not work, bandwidth exceeded.
EXPECT_CALL(mock_paced_sender_, EnqueuePackets).Times(0);
rtp_sender_->OnReceivedNack(sequence_numbers, 0);
}
TEST_F(RtpSenderTest, UpdatingCsrcsUpdatedOverhead) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
config.rtx_send_ssrc = {};
CreateSender(config);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
// Using packet with two csrcs adds 2*4 bytes to the header.
uint32_t csrcs[] = {1, 2};
rtp_sender_->AllocatePacket(csrcs);
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 20u);
}
TEST_F(RtpSenderTest, OnOverheadChanged) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
config.rtx_send_ssrc = {};
CreateSender(config);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
rtp_sender_->RegisterRtpHeaderExtension(TransmissionOffset::Uri(),
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_F(RtpSenderTest, CountMidOnlyUntilAcked) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
config.rtx_send_ssrc = {};
CreateSender(config);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
rtp_sender_->RegisterRtpHeaderExtension(RtpMid::Uri(), kMidExtensionId);
// Counted only if set.
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
rtp_sender_->SetMid("foo");
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 36u);
rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::Uri(), kRidExtensionId);
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 52u);
// Ack received, mid/rid no longer sent.
rtp_sender_->OnReceivedAckOnSsrc(0);
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
}
TEST_F(RtpSenderTest, CountMidRidRridUntilAcked) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
CreateSender(config);
// Base RTP overhead is 12B and we use RTX which has an additional 2 bytes
// overhead.
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 14u);
rtp_sender_->RegisterRtpHeaderExtension(RtpMid::Uri(), kMidExtensionId);
// Counted only if set.
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 14u);
rtp_sender_->SetMid("foo");
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 38u);
rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::Uri(), kRidExtensionId);
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 54u);
// mid/rrid may be shared with mid/rid when both are active.
rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(),
kRepairedRidExtensionId);
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 54u);
// Ack received, mid/rid no longer sent but we still need space for
// mid/rrid which can no longer be shared with mid/rid.
rtp_sender_->OnReceivedAckOnSsrc(0);
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 54u);
// Ack received for RTX, no need to send RRID anymore.
rtp_sender_->OnReceivedAckOnRtxSsrc(0);
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 14u);
}
TEST_F(RtpSenderTest, DontCountVolatileExtensionsIntoOverhead) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
config.rtx_send_ssrc = {};
CreateSender(config);
// Base RTP overhead is 12B.
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
rtp_sender_->RegisterRtpHeaderExtension(InbandComfortNoiseExtension::Uri(),
1);
rtp_sender_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
2);
rtp_sender_->RegisterRtpHeaderExtension(VideoOrientation::Uri(), 3);
rtp_sender_->RegisterRtpHeaderExtension(PlayoutDelayLimits::Uri(), 4);
rtp_sender_->RegisterRtpHeaderExtension(VideoContentTypeExtension::Uri(), 5);
rtp_sender_->RegisterRtpHeaderExtension(VideoTimingExtension::Uri(), 6);
rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(), 7);
rtp_sender_->RegisterRtpHeaderExtension(ColorSpaceExtension::Uri(), 8);
// Still only 12B counted since can't count on above being sent.
EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
}
TEST_F(RtpSenderTest, SendPacketHandlesRetransmissionHistory) {
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, env_.clock().CurrentTime());
const uint16_t media_sequence_number = packet->SequenceNumber();
packet->set_allow_retransmission(true);
packet_history_->PutRtpPacket(std::move(packet), env_.clock().CurrentTime());
// 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.
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_F(RtpSenderTest, MarksRetransmittedPackets) {
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, env_.clock().CurrentTime());
const uint32_t kMediaSsrc = 567;
const uint16_t kMediaSequenceNumber = 123;
packet->SetSsrc(kMediaSsrc);
packet->SetSequenceNumber(kMediaSequenceNumber);
packet->set_allow_retransmission(true);
packet_history_->PutRtpPacket(std::move(packet), env_.clock().CurrentTime());
// 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::original_ssrc, kMediaSsrc)),
Pointee(Property(&RtpPacketToSend::retransmitted_sequence_number,
Eq(kMediaSequenceNumber)))))));
EXPECT_THAT(rtp_sender_->ReSendPacket(kMediaSequenceNumber), Gt(0));
}
TEST_F(RtpSenderTest, GeneratedPaddingHasBweExtensions) {
// Min requested size in order to use RTX payload.
const size_t kMinPaddingSize = 50;
EnableRtx();
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransmissionOffset::Uri(), kTransmissionTimeOffsetExtensionId));
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
// Put a packet in the history, in order to facilitate payload padding.
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, env_.clock().CurrentTime());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kMinPaddingSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet_history_->PutRtpPacket(std::move(packet), env_.clock().CurrentTime());
// Generate a plain padding packet, check that extensions are registered.
std::vector<std::unique_ptr<RtpPacketToSend>> generated_packets =
GeneratePadding(/*target_size_bytes=*/1);
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>());
EXPECT_GT(plain_padding->padding_size(), 0u);
// Generate a payload padding packets, check that extensions are registered.
generated_packets = GeneratePadding(kMinPaddingSize);
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>());
EXPECT_GT(payload_padding->payload_size(), 0u);
}
TEST_F(RtpSenderTest, GeneratedPaddingHasMidRidExtensions) {
EnableMidSending("mid");
EnableRidSending();
// 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.
packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, env_.clock().CurrentTime());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(1234);
packet->set_packet_type(RtpPacketMediaType::kVideo);
sequencer_->Sequence(*packet);
packet_history_->PutRtpPacket(std::move(packet), env_.clock().CurrentTime());
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
GeneratePadding(/*target_size_bytes=*/1);
ASSERT_THAT(padding_packets, SizeIs(1));
EXPECT_TRUE(padding_packets[0]->HasExtension<RtpMid>());
EXPECT_TRUE(padding_packets[0]->HasExtension<RtpStreamId>());
}
TEST_F(RtpSenderTest, GeneratedPaddingOnRtxHasMidRidExtensions) {
EnableRtx();
EnableMidSending("mid");
EnableRidSending();
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
GeneratePadding(/*target_size_bytes=*/1);
ASSERT_THAT(padding_packets, SizeIs(1));
EXPECT_TRUE(padding_packets[0]->HasExtension<RtpMid>());
EXPECT_TRUE(padding_packets[0]->HasExtension<RepairedRtpStreamId>());
}
TEST_F(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
// Min requested size in order to use RTX payload.
const size_t kMinPaddingSize = 50;
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
const size_t kPayloadPacketSize = kMinPaddingSize;
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, env_.clock().CurrentTime());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet_history_->PutRtpPacket(std::move(packet), env_.clock().CurrentTime());
// Generated padding has large enough budget that the video packet should be
// retransmitted as padding.
std::vector<std::unique_ptr<RtpPacketToSend>> generated_packets =
GeneratePadding(kMinPaddingSize);
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 = GeneratePadding(kPaddingBytesRequested);
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, kMaxPaddingLength);
}
TEST_F(RtpSenderTest, LimitsPayloadPaddingSize) {
// RTX payload padding is limited to 3x target size.
const double kFactor = 3.0;
SetUpRtpSender(false, false, nullptr);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::Uri(), 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, env_.clock().CurrentTime());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
packet_history_->PutRtpPacket(std::move(packet), env_.clock().CurrentTime());
// 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(
GeneratePadding(kMinTargerSizeForPayload),
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(
GeneratePadding(kMinTargerSizeForPayload - 1),
AllOf(Not(IsEmpty()),
Each(Pointee(Property(&RtpPacketToSend::padding_size, Gt(0u))))));
}
TEST_F(RtpSenderTest, GeneratePaddingCreatesPurePaddingWithoutRtx) {
packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransmissionOffset::Uri(), kTransmissionTimeOffsetExtensionId));
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::Uri(), 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, env_.clock().CurrentTime());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
sequencer_->Sequence(*packet);
packet_history_->PutRtpPacket(std::move(packet), env_.clock().CurrentTime());
// 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 + kMaxPaddingLength - 1) / kMaxPaddingLength;
size_t padding_bytes_generated = 0;
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
GeneratePadding(kPaddingBytesRequested);
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>());
}
EXPECT_EQ(padding_bytes_generated,
kExpectedNumPaddingPackets * kMaxPaddingLength);
}
TEST_F(RtpSenderTest, SupportsPadding) {
bool kSendingMediaStats[] = {true, false};
bool kEnableRedundantPayloads[] = {true, false};
absl::string_view kBweExtensionUris[] = {
TransportSequenceNumber::Uri(), TransportSequenceNumberV2::Uri(),
AbsoluteSendTime::Uri(), TransmissionOffset::Uri()};
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_->SetRtxPayloadType(kRtxPayload, kPayload);
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_F(RtpSenderTest, SetsCaptureTimeOnRtxRetransmissions) {
EnableRtx();
// Put a packet in the packet history, with current time as capture time.
const Timestamp start_time = env_.clock().CurrentTime();
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, kMarkerBit, /*rtp_timestamp=*/0,
/*capture_time=*/start_time);
packet->set_allow_retransmission(true);
sequencer_->Sequence(*packet);
packet_history_->PutRtpPacket(std::move(packet), start_time);
// 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, start_time)))));
EXPECT_GT(rtp_sender_->ReSendPacket(kSeqNum), 0);
}
TEST_F(RtpSenderTest, IgnoresNackAfterDisablingMedia) {
const TimeDelta kRtt = TimeDelta::Millis(10);
EnableRtx();
packet_history_->SetRtt(kRtt);
// Put a packet in the history.
const Timestamp start_time = env_.clock().CurrentTime();
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, kMarkerBit, 0, /*capture_time=*/start_time);
packet->set_allow_retransmission(true);
sequencer_->Sequence(*packet);
packet_history_->PutRtpPacket(std::move(packet), start_time);
// Disable media sending and try to retransmit the packet, it should fail.
rtp_sender_->SetSendingMediaStatus(false);
time_controller_.AdvanceTime(kRtt);
EXPECT_LT(rtp_sender_->ReSendPacket(kSeqNum), 0);
}
TEST_F(RtpSenderTest, DoesntFecProtectRetransmissions) {
// Set up retranmission without RTX, so that a plain copy of the old packet is
// re-sent instead.
const TimeDelta kRtt = TimeDelta::Millis(10);
rtp_sender_->SetSendingMediaStatus(true);
rtp_sender_->SetRtxStatus(kRtxOff);
packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
packet_history_->SetRtt(kRtt);
// Put a fec protected packet in the history.
const Timestamp start_time = env_.clock().CurrentTime();
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, kMarkerBit, 0, start_time);
packet->set_allow_retransmission(true);
packet->set_fec_protect_packet(true);
sequencer_->Sequence(*packet);
packet_history_->PutRtpPacket(std::move(packet), start_time);
// 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(kRtt);
EXPECT_GT(rtp_sender_->ReSendPacket(kSeqNum), 0);
}
TEST_F(RtpSenderTest, MarksPacketsWithKeyframeStatus) {
RTPSenderVideo::Config video_config;
video_config.clock = &env_.clock();
video_config.rtp_sender = rtp_sender_.get();
video_config.field_trials = &env_.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;
Timestamp capture_time = env_.clock().CurrentTime();
EXPECT_TRUE(rtp_sender_video.SendVideo(
kPayloadType, kCodecType,
capture_time.ms() * kCaptureTimeMsToRtpTimestamp, capture_time,
kPayloadData, sizeof(kPayloadData), video_header,
kDefaultExpectedRetransmissionTime, {}));
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;
Timestamp capture_time = env_.clock().CurrentTime();
EXPECT_TRUE(rtp_sender_video.SendVideo(
kPayloadType, kCodecType,
capture_time.ms() * kCaptureTimeMsToRtpTimestamp, capture_time,
kPayloadData, sizeof(kPayloadData), video_header,
kDefaultExpectedRetransmissionTime, {}));
time_controller_.AdvanceTime(TimeDelta::Millis(33));
}
}
} // namespace webrtc
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