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:
@ -34,6 +34,7 @@
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#include "modules/rtp_rtcp/source/rtp_utility.h"
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#include "rtc_base/arraysize.h"
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#include "rtc_base/rate_limiter.h"
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#include "rtc_base/strings/string_builder.h"
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#include "test/field_trial.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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@ -75,12 +76,18 @@ const char kNoMid[] = "";
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using ::testing::_;
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using ::testing::AllOf;
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using ::testing::Contains;
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using ::testing::Each;
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using ::testing::ElementsAreArray;
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using ::testing::Eq;
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using ::testing::Field;
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using ::testing::Gt;
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using ::testing::IsEmpty;
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using ::testing::NiceMock;
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using ::testing::Not;
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using ::testing::Pointee;
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using ::testing::Property;
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using ::testing::Return;
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using ::testing::SizeIs;
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using ::testing::StrictMock;
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uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
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@ -140,14 +147,6 @@ struct TestConfig {
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bool with_overhead = false;
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};
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std::string ToFieldTrialString(TestConfig config) {
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std::string field_trials;
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if (config.with_overhead) {
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field_trials += "WebRTC-SendSideBwe-WithOverhead/Enabled/";
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}
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return field_trials;
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}
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class MockRtpPacketPacer : public RtpPacketSender {
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public:
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MockRtpPacketPacer() {}
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@ -236,6 +235,31 @@ struct RtpSenderContext {
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RTPSender packet_generator_;
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};
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class FieldTrialConfig : public WebRtcKeyValueConfig {
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public:
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FieldTrialConfig() : overhead_enabled_(false), max_padding_factor_(1200) {}
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~FieldTrialConfig() override {}
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void SetOverHeadEnabled(bool enabled) { overhead_enabled_ = enabled; }
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void SetMaxPaddingFactor(double factor) { max_padding_factor_ = factor; }
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std::string Lookup(absl::string_view key) const override {
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if (key == "WebRTC-LimitPaddingSize") {
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char string_buf[32];
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rtc::SimpleStringBuilder ssb(string_buf);
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ssb << "factor:" << max_padding_factor_;
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return ssb.str();
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} else if (key == "WebRTC-SendSideBwe-WithOverhead") {
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return overhead_enabled_ ? "Enabled" : "Disabled";
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}
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return "";
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}
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private:
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bool overhead_enabled_;
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double max_padding_factor_;
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};
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} // namespace
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class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
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@ -251,8 +275,9 @@ class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
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std::vector<RtpExtensionSize>(),
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nullptr,
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&fake_clock_),
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kMarkerBit(true),
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field_trials_(ToFieldTrialString(GetParam())) {}
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kMarkerBit(true) {
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field_trials_.SetOverHeadEnabled(GetParam().with_overhead);
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}
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void SetUp() override { SetUpRtpSender(true, false, false); }
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@ -282,6 +307,8 @@ class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
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config.populate_network2_timestamp = populate_network2;
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config.rtp_stats_callback = &rtp_stats_callback_;
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config.always_send_mid_and_rid = always_send_mid_and_rid;
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config.field_trials = &field_trials_;
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rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
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rtp_sender()->SetSequenceNumber(kSeqNum);
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rtp_sender()->SetTimestampOffset(0);
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@ -299,7 +326,7 @@ class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
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LoopbackTransportTest transport_;
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const bool kMarkerBit;
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test::ScopedFieldTrials field_trials_;
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FieldTrialConfig field_trials_;
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StreamDataTestCallback rtp_stats_callback_;
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std::unique_ptr<RtpPacketToSend> BuildRtpPacket(int payload_type,
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@ -522,6 +549,7 @@ TEST_P(RtpSenderTestWithoutPacer,
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config.event_log = &mock_rtc_event_log_;
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config.retransmission_rate_limiter = &retransmission_rate_limiter_;
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config.overhead_observer = &mock_overhead_observer;
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config.field_trials = &field_trials_;
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rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
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EXPECT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
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@ -2243,7 +2271,7 @@ TEST_P(RtpSenderTest, SendPacketUpdatesStats) {
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EXPECT_EQ(rtx_stats.retransmitted.packets, 1u);
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}
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TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
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TEST_P(RtpSenderTest, GeneratedPaddingHasBweExtensions) {
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// Min requested size in order to use RTX payload.
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const size_t kMinPaddingSize = 50;
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@ -2262,7 +2290,73 @@ TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
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kRtpExtensionTransportSequenceNumber,
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kTransportSequenceNumberExtensionId));
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const size_t kPayloadPacketSize = 1234;
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// Send a payload packet first, to enable padding and populate the packet
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// history.
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std::unique_ptr<RtpPacketToSend> packet =
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BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
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packet->set_allow_retransmission(true);
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packet->SetPayloadSize(kMinPaddingSize);
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packet->set_packet_type(RtpPacketMediaType::kVideo);
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EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
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rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
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// Generate a plain padding packet, check that extensions are registered.
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std::vector<std::unique_ptr<RtpPacketToSend>> generated_packets =
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rtp_sender()->GeneratePadding(/*target_size_bytes=*/1, true);
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ASSERT_THAT(generated_packets, SizeIs(1));
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auto& plain_padding = generated_packets.front();
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EXPECT_GT(plain_padding->padding_size(), 0u);
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EXPECT_TRUE(plain_padding->HasExtension<TransportSequenceNumber>());
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EXPECT_TRUE(plain_padding->HasExtension<AbsoluteSendTime>());
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EXPECT_TRUE(plain_padding->HasExtension<TransmissionOffset>());
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// Verify all header extensions have been written.
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rtp_egress()->SendPacket(plain_padding.get(), PacedPacketInfo());
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const auto& sent_plain_padding = transport_.last_sent_packet();
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EXPECT_TRUE(sent_plain_padding.HasExtension<TransportSequenceNumber>());
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EXPECT_TRUE(sent_plain_padding.HasExtension<AbsoluteSendTime>());
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EXPECT_TRUE(sent_plain_padding.HasExtension<TransmissionOffset>());
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webrtc::RTPHeader rtp_header;
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sent_plain_padding.GetHeader(&rtp_header);
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EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
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EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
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EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
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// Generate a payload padding packets, check that extensions are registered.
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generated_packets = rtp_sender()->GeneratePadding(kMinPaddingSize, true);
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ASSERT_EQ(generated_packets.size(), 1u);
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auto& payload_padding = generated_packets.front();
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EXPECT_EQ(payload_padding->padding_size(), 0u);
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EXPECT_TRUE(payload_padding->HasExtension<TransportSequenceNumber>());
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EXPECT_TRUE(payload_padding->HasExtension<AbsoluteSendTime>());
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EXPECT_TRUE(payload_padding->HasExtension<TransmissionOffset>());
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// Verify all header extensions have been written.
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rtp_egress()->SendPacket(payload_padding.get(), PacedPacketInfo());
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const auto& sent_payload_padding = transport_.last_sent_packet();
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EXPECT_TRUE(sent_payload_padding.HasExtension<TransportSequenceNumber>());
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EXPECT_TRUE(sent_payload_padding.HasExtension<AbsoluteSendTime>());
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EXPECT_TRUE(sent_payload_padding.HasExtension<TransmissionOffset>());
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sent_payload_padding.GetHeader(&rtp_header);
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EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
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EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
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EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
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}
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TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
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// Min requested size in order to use RTX payload.
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const size_t kMinPaddingSize = 50;
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rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
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rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload);
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rtp_sender_context_->packet_history_.SetStorePacketsStatus(
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RtpPacketHistory::StorageMode::kStoreAndCull, 1);
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ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
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kRtpExtensionTransportSequenceNumber,
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kTransportSequenceNumberExtensionId));
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const size_t kPayloadPacketSize = kMinPaddingSize;
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std::unique_ptr<RtpPacketToSend> packet =
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BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
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packet->set_allow_retransmission(true);
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@ -2283,17 +2377,6 @@ TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
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EXPECT_EQ(padding_packet->Ssrc(), kRtxSsrc);
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EXPECT_EQ(padding_packet->payload_size(),
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kPayloadPacketSize + kRtxHeaderSize);
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EXPECT_TRUE(padding_packet->HasExtension<TransportSequenceNumber>());
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EXPECT_TRUE(padding_packet->HasExtension<AbsoluteSendTime>());
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EXPECT_TRUE(padding_packet->HasExtension<TransmissionOffset>());
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// Verify all header extensions are received.
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rtp_egress()->SendPacket(padding_packet.get(), PacedPacketInfo());
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webrtc::RTPHeader rtp_header;
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transport_.last_sent_packet().GetHeader(&rtp_header);
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EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
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EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
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EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
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// Not enough budged for payload padding, use plain padding instead.
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const size_t kPaddingBytesRequested = kMinPaddingSize - 1;
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@ -2308,23 +2391,55 @@ TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
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EXPECT_EQ(packet->payload_size(), 0u);
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EXPECT_GT(packet->padding_size(), 0u);
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padding_bytes_generated += packet->padding_size();
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EXPECT_TRUE(packet->HasExtension<TransportSequenceNumber>());
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EXPECT_TRUE(packet->HasExtension<AbsoluteSendTime>());
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EXPECT_TRUE(packet->HasExtension<TransmissionOffset>());
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// Verify all header extensions are received.
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rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
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webrtc::RTPHeader rtp_header;
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transport_.last_sent_packet().GetHeader(&rtp_header);
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EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
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EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
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EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
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}
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EXPECT_EQ(padding_bytes_generated, kMaxPaddingSize);
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}
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TEST_P(RtpSenderTest, LimitsPayloadPaddingSize) {
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// Limit RTX payload padding to 2x target size.
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const double kFactor = 2.0;
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field_trials_.SetMaxPaddingFactor(kFactor);
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SetUpRtpSender(true, false, false);
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rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
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rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload);
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rtp_sender_context_->packet_history_.SetStorePacketsStatus(
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RtpPacketHistory::StorageMode::kStoreAndCull, 1);
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ASSERT_EQ(0, rtp_sender()->RegisterRtpHeaderExtension(
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kRtpExtensionTransportSequenceNumber,
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kTransportSequenceNumberExtensionId));
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// Send a dummy video packet so it ends up in the packet history.
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const size_t kPayloadPacketSize = 1234u;
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std::unique_ptr<RtpPacketToSend> packet =
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BuildRtpPacket(kPayload, true, 0, fake_clock_.TimeInMilliseconds());
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packet->set_allow_retransmission(true);
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packet->SetPayloadSize(kPayloadPacketSize);
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packet->set_packet_type(RtpPacketMediaType::kVideo);
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EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
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rtp_egress()->SendPacket(packet.get(), PacedPacketInfo());
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// Smallest target size that will result in the sent packet being returned as
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// padding.
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const size_t kMinTargerSizeForPayload =
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(kPayloadPacketSize + kRtxHeaderSize) / kFactor;
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// Generated padding has large enough budget that the video packet should be
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// retransmitted as padding.
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EXPECT_THAT(
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rtp_sender()->GeneratePadding(kMinTargerSizeForPayload, true),
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AllOf(Not(IsEmpty()),
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Each(Pointee(Property(&RtpPacketToSend::padding_size, Eq(0u))))));
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// If payload padding is > 2x requested size, plain padding is returned
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// instead.
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EXPECT_THAT(
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rtp_sender()->GeneratePadding(kMinTargerSizeForPayload - 1, true),
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AllOf(Not(IsEmpty()),
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Each(Pointee(Property(&RtpPacketToSend::padding_size, Gt(0u))))));
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}
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TEST_P(RtpSenderTest, GeneratePaddingCreatesPurePaddingWithoutRtx) {
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rtp_sender_context_->packet_history_.SetStorePacketsStatus(
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RtpPacketHistory::StorageMode::kStoreAndCull, 1);
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