Limits size of payload padding packets to 2x target size.

This CL also slightly refactors unit test, to test fewer things each.

Bug: webrtc:11508
Change-Id: I98553d2b185364132c626d373683f38891e36c6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31087}
This commit is contained in:
Erik Språng
2020-04-16 15:07:56 +02:00
committed by Commit Bot
parent b33a0ca1ee
commit e886d2ebc3
5 changed files with 180 additions and 37 deletions

View File

@ -34,6 +34,7 @@
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/strings/string_builder.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
@ -75,12 +76,18 @@ const char kNoMid[] = "";
using ::testing::_;
using ::testing::AllOf;
using ::testing::Contains;
using ::testing::Each;
using ::testing::ElementsAreArray;
using ::testing::Eq;
using ::testing::Field;
using ::testing::Gt;
using ::testing::IsEmpty;
using ::testing::NiceMock;
using ::testing::Not;
using ::testing::Pointee;
using ::testing::Property;
using ::testing::Return;
using ::testing::SizeIs;
using ::testing::StrictMock;
uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
@ -140,14 +147,6 @@ struct TestConfig {
bool with_overhead = false;
};
std::string ToFieldTrialString(TestConfig config) {
std::string field_trials;
if (config.with_overhead) {
field_trials += "WebRTC-SendSideBwe-WithOverhead/Enabled/";
}
return field_trials;
}
class MockRtpPacketPacer : public RtpPacketSender {
public:
MockRtpPacketPacer() {}
@ -236,6 +235,31 @@ struct RtpSenderContext {
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> {
@ -251,8 +275,9 @@ class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
std::vector<RtpExtensionSize>(),
nullptr,
&fake_clock_),
kMarkerBit(true),
field_trials_(ToFieldTrialString(GetParam())) {}
kMarkerBit(true) {
field_trials_.SetOverHeadEnabled(GetParam().with_overhead);
}
void SetUp() override { SetUpRtpSender(true, false, false); }
@ -282,6 +307,8 @@ class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
config.populate_network2_timestamp = populate_network2;
config.rtp_stats_callback = &rtp_stats_callback_;
config.always_send_mid_and_rid = always_send_mid_and_rid;
config.field_trials = &field_trials_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
rtp_sender()->SetSequenceNumber(kSeqNum);
rtp_sender()->SetTimestampOffset(0);
@ -299,7 +326,7 @@ class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
LoopbackTransportTest transport_;
const bool kMarkerBit;
test::ScopedFieldTrials field_trials_;
FieldTrialConfig field_trials_;
StreamDataTestCallback rtp_stats_callback_;
std::unique_ptr<RtpPacketToSend> BuildRtpPacket(int payload_type,
@ -522,6 +549,7 @@ TEST_P(RtpSenderTestWithoutPacer,
config.event_log = &mock_rtc_event_log_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.overhead_observer = &mock_overhead_observer;
config.field_trials = &field_trials_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
@ -2243,7 +2271,7 @@ TEST_P(RtpSenderTest, SendPacketUpdatesStats) {
EXPECT_EQ(rtx_stats.retransmitted.packets, 1u);
}
TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
TEST_P(RtpSenderTest, GeneratedPaddingHasBweExtensions) {
// Min requested size in order to use RTX payload.
const size_t kMinPaddingSize = 50;
@ -2262,7 +2290,73 @@ TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
const size_t kPayloadPacketSize = 1234;
// Send a payload packet first, to enable padding and populate the packet
// history.
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kMinPaddingSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Generate a plain padding packet, check that extensions are registered.
std::vector<std::unique_ptr<RtpPacketToSend>> generated_packets =
rtp_sender()->GeneratePadding(/*target_size_bytes=*/1, true);
ASSERT_THAT(generated_packets, SizeIs(1));
auto& plain_padding = generated_packets.front();
EXPECT_GT(plain_padding->padding_size(), 0u);
EXPECT_TRUE(plain_padding->HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(plain_padding->HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(plain_padding->HasExtension<TransmissionOffset>());
// Verify all header extensions have been written.
rtp_egress()->SendPacket(plain_padding.get(), PacedPacketInfo());
const auto& sent_plain_padding = transport_.last_sent_packet();
EXPECT_TRUE(sent_plain_padding.HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(sent_plain_padding.HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(sent_plain_padding.HasExtension<TransmissionOffset>());
webrtc::RTPHeader rtp_header;
sent_plain_padding.GetHeader(&rtp_header);
EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
// Generate a payload padding packets, check that extensions are registered.
generated_packets = rtp_sender()->GeneratePadding(kMinPaddingSize, true);
ASSERT_EQ(generated_packets.size(), 1u);
auto& payload_padding = generated_packets.front();
EXPECT_EQ(payload_padding->padding_size(), 0u);
EXPECT_TRUE(payload_padding->HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(payload_padding->HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(payload_padding->HasExtension<TransmissionOffset>());
// Verify all header extensions have been written.
rtp_egress()->SendPacket(payload_padding.get(), PacedPacketInfo());
const auto& sent_payload_padding = transport_.last_sent_packet();
EXPECT_TRUE(sent_payload_padding.HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(sent_payload_padding.HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(sent_payload_padding.HasExtension<TransmissionOffset>());
sent_payload_padding.GetHeader(&rtp_header);
EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
}
TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
// Min requested size in order to use RTX payload.
const size_t kMinPaddingSize = 50;
rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
const size_t kPayloadPacketSize = kMinPaddingSize;
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_allow_retransmission(true);
@ -2283,17 +2377,6 @@ TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
EXPECT_EQ(padding_packet->Ssrc(), kRtxSsrc);
EXPECT_EQ(padding_packet->payload_size(),
kPayloadPacketSize + kRtxHeaderSize);
EXPECT_TRUE(padding_packet->HasExtension<TransportSequenceNumber>());
EXPECT_TRUE(padding_packet->HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(padding_packet->HasExtension<TransmissionOffset>());
// Verify all header extensions are received.
rtp_egress()->SendPacket(padding_packet.get(), 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);
// Not enough budged for payload padding, use plain padding instead.
const size_t kPaddingBytesRequested = kMinPaddingSize - 1;
@ -2308,23 +2391,55 @@ TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
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_egress()->SendPacket(packet.get(), 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, kMaxPaddingSize);
}
TEST_P(RtpSenderTest, LimitsPayloadPaddingSize) {
// Limit RTX payload padding to 2x target size.
const double kFactor = 2.0;
field_trials_.SetMaxPaddingFactor(kFactor);
SetUpRtpSender(true, false, false);
rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
// Send a dummy video packet so it ends up in the packet history.
const size_t kPayloadPacketSize = 1234u;
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
// Smallest target size that will result in the sent packet being returned as
// padding.
const size_t kMinTargerSizeForPayload =
(kPayloadPacketSize + kRtxHeaderSize) / kFactor;
// Generated padding has large enough budget that the video packet should be
// retransmitted as padding.
EXPECT_THAT(
rtp_sender()->GeneratePadding(kMinTargerSizeForPayload, true),
AllOf(Not(IsEmpty()),
Each(Pointee(Property(&RtpPacketToSend::padding_size, Eq(0u))))));
// If payload padding is > 2x requested size, plain padding is returned
// instead.
EXPECT_THAT(
rtp_sender()->GeneratePadding(kMinTargerSizeForPayload - 1, true),
AllOf(Not(IsEmpty()),
Each(Pointee(Property(&RtpPacketToSend::padding_size, Gt(0u))))));
}
TEST_P(RtpSenderTest, GeneratePaddingCreatesPurePaddingWithoutRtx) {
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);