Clean up RtpSenderTest and remove RtpSenderEgress dependencies.
Since all test cases that used RtpSenderEgress have been refactored or
moved, we can now get rid of lot of test fixture crud:
* Remove RtpSenderContext helper, make sender normal member.
* Remove test transport helper
* Remove task queue helper (needed for thread checks in egress)
* Remove various mocks no longer used
* Remove RtpSenderWithoutPacer subclass
* Remove WithWithoutOverhead parametrization (only affect egress)
..plus some cleanup of how configs are created.
Bug: webrtc:11340
Change-Id: I5c581d60862fc6dc2b99f76058782309dc7aef4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220280
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34135}
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index b33dcd9..e9be0161 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -22,14 +22,12 @@
#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"
@@ -87,64 +85,10 @@
using ::testing::IsEmpty;
using ::testing::NiceMock;
using ::testing::Not;
-using ::testing::Optional;
using ::testing::Pointee;
using ::testing::Property;
using ::testing::Return;
using ::testing::SizeIs;
-using ::testing::StrEq;
-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:
@@ -157,102 +101,11 @@
(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() : 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 {
@@ -261,20 +114,17 @@
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> {
+class RtpSenderTest : public ::testing::Test {
protected:
RtpSenderTest()
: time_controller_(Timestamp::Millis(kStartTime)),
@@ -289,64 +139,51 @@
nullptr,
clock_),
kMarkerBit(true) {
- field_trials_.SetOverHeadEnabled(GetParam().with_overhead);
}
- void SetUp() override { SetUpRtpSender(true, false, false); }
+ void SetUp() override { SetUpRtpSender(true, false, nullptr); }
- 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,
+ void SetUpRtpSender(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;
+ RtpRtcpInterface::Configuration config = GetDefaultConfig();
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_;
+ CreateSender(config);
+ }
- rtp_sender_context_ =
- std::make_unique<RtpSenderContext>(config, &time_controller_);
- rtp_sender()->SetSequenceNumber(kSeqNum);
- rtp_sender()->SetTimestampOffset(0);
+ RtpRtcpInterface::Configuration GetDefaultConfig() {
+ RtpRtcpInterface::Configuration config;
+ config.clock = 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 = &field_trials_;
+ return config;
+ }
+
+ void CreateSender(const RtpRtcpInterface::Configuration& config) {
+ packet_history_ = std::make_unique<RtpPacketHistory>(
+ config.clock, config.enable_rtx_padding_prioritization);
+ rtp_sender_ = std::make_unique<RTPSender>(config, packet_history_.get(),
+ config.paced_sender);
+ 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_;
+ std::unique_ptr<RtpPacketHistory> packet_history_;
+ std::unique_ptr<RTPSender> rtp_sender_;
- LoopbackTransportTest transport_;
const bool kMarkerBit;
FieldTrialConfig field_trials_;
@@ -354,13 +191,13 @@
bool marker_bit,
uint32_t timestamp,
int64_t capture_time_ms) {
- auto packet = rtp_sender()->AllocatePacket();
+ 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()));
+ EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
return packet;
}
@@ -373,8 +210,8 @@
packet->set_allow_retransmission(true);
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender()->SendToNetwork(
- std::make_unique<RtpPacketToSend>(*packet)));
+ EXPECT_TRUE(
+ rtp_sender_->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
return packet;
}
@@ -384,16 +221,15 @@
// 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());
+ /*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)) {
+ for (auto& packet : rtp_sender_->GeneratePadding(target_size_bytes, true)) {
generated_bytes += packet->payload_size() + packet->padding_size();
- rtp_sender()->SendToNetwork(std::move(packet));
+ rtp_sender_->SendToNetwork(std::move(packet));
}
return generated_bytes;
}
@@ -406,64 +242,56 @@
// 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(
+ packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, kNumberOfPacketsToStore);
- rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload);
- rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+ 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);
+ 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);
+ 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) {
+TEST_F(RtpSenderTest, AllocatePacketSetCsrc) {
// Configure rtp_sender with csrc.
std::vector<uint32_t> csrcs;
csrcs.push_back(0x23456789);
- rtp_sender()->SetCsrcs(csrcs);
+ rtp_sender_->SetCsrcs(csrcs);
- auto packet = rtp_sender()->AllocatePacket();
+ auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet);
- EXPECT_EQ(rtp_sender()->SSRC(), packet->Ssrc());
+ EXPECT_EQ(rtp_sender_->SSRC(), packet->Ssrc());
EXPECT_EQ(csrcs, packet->Csrcs());
}
-TEST_P(RtpSenderTestWithoutPacer, AllocatePacketReserveExtensions) {
+TEST_F(RtpSenderTest, AllocatePacketReserveExtensions) {
// Configure rtp_sender with extensions.
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransmissionOffset::kUri, kTransmissionTimeOffsetExtensionId));
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(AudioLevel::kUri,
- kAudioLevelExtensionId));
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(AudioLevel::kUri,
+ kAudioLevelExtensionId));
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
VideoOrientation::kUri, kVideoRotationExtensionId));
- auto packet = rtp_sender()->AllocatePacket();
+ auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet);
// Preallocate BWE extensions RtpSender set itself.
@@ -475,27 +303,16 @@
EXPECT_FALSE(packet->HasExtension<VideoOrientation>());
}
-TEST_P(RtpSenderTest, PaddingAlwaysAllowedOnAudio) {
- MockTransport transport;
- RtpRtcpInterface::Configuration config;
+TEST_F(RtpSenderTest, PaddingAlwaysAllowedOnAudio) {
+ RtpRtcpInterface::Configuration config = GetDefaultConfig();
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_);
+ CreateSender(config);
- rtp_sender()->SetTimestampOffset(0);
-
- std::unique_ptr<RtpPacketToSend> audio_packet =
- rtp_sender()->AllocatePacket();
+ 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());
+ rtp_sender_->AssignSequenceNumber(audio_packet.get());
const size_t kPaddingSize = 59;
@@ -518,7 +335,7 @@
EXPECT_EQ(kMinPaddingSize, GenerateAndSendPadding(kMinPaddingSize - 5));
}
-TEST_P(RtpSenderTest, SendToNetworkForwardsPacketsToPacer) {
+TEST_F(RtpSenderTest, SendToNetworkForwardsPacketsToPacer) {
auto packet = BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, 0);
int64_t now_ms = clock_->TimeInMilliseconds();
@@ -529,17 +346,17 @@
Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum)),
Pointee(Property(&RtpPacketToSend::capture_time_ms, now_ms))))));
EXPECT_TRUE(
- rtp_sender()->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
+ rtp_sender_->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
}
-TEST_P(RtpSenderTest, ReSendPacketForwardsPacketsToPacer) {
- rtp_sender_context_->packet_history_.SetStorePacketsStatus(
+TEST_F(RtpSenderTest, ReSendPacketForwardsPacketsToPacer) {
+ 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);
+ packet_history_->PutRtpPacket(std::move(packet), now_ms);
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(AllOf(
@@ -548,12 +365,12 @@
Pointee(Property(&RtpPacketToSend::capture_time_ms, now_ms)),
Pointee(Property(&RtpPacketToSend::packet_type,
RtpPacketMediaType::kRetransmission))))));
- EXPECT_TRUE(rtp_sender()->ReSendPacket(seq_no));
+ 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) {
+TEST_F(RtpSenderTest, SendPadding) {
constexpr int kNumPaddingPackets = 4;
EXPECT_CALL(mock_paced_sender_, EnqueuePackets);
std::unique_ptr<RtpPacketToSend> media_packet =
@@ -578,10 +395,10 @@
Pointee(Property(&RtpPacketToSend::Timestamp,
media_packet->Timestamp()))))));
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
- rtp_sender()->GeneratePadding(kPaddingTargetBytes,
- /*media_has_been_sent=*/true);
+ rtp_sender_->GeneratePadding(kPaddingTargetBytes,
+ /*media_has_been_sent=*/true);
ASSERT_THAT(padding_packets, SizeIs(1));
- rtp_sender()->SendToNetwork(std::move(padding_packets[0]));
+ rtp_sender_->SendToNetwork(std::move(padding_packets[0]));
}
// Send a regular video packet again.
@@ -598,53 +415,53 @@
/*payload_size=*/100);
}
-TEST_P(RtpSenderTest, NoPaddingAsFirstPacketWithoutBweExtensions) {
- EXPECT_THAT(rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
- /*media_has_been_sent=*/false),
+TEST_F(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),
+ EXPECT_THAT(rtp_sender_->GeneratePadding(/*target_size_bytes=*/100,
+ /*media_has_been_sent=*/false),
IsEmpty());
}
-TEST_P(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithTransportCc) {
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+TEST_F(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),
+ 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),
+ 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(
+TEST_F(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),
+ 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),
+ EXPECT_THAT(rtp_sender_->GeneratePadding(/*target_size_bytes=*/100,
+ /*media_has_been_sent=*/false),
Not(IsEmpty()));
}
-TEST_P(RtpSenderTest, UpdatesTimestampsOnPlainRtxPadding) {
+TEST_F(RtpSenderTest, UpdatesTimestampsOnPlainRtxPadding) {
EnableRtx();
// Timestamps as set based on capture time in RtpSenderTest.
const int64_t start_time = clock_->TimeInMilliseconds();
@@ -666,8 +483,8 @@
// Timestamps on padding should be offset from the sent media.
EXPECT_THAT(
- rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
- /*media_has_been_sent=*/true),
+ rtp_sender_->GeneratePadding(/*target_size_bytes=*/100,
+ /*media_has_been_sent=*/true),
Each(AllOf(
Pointee(Property(&RtpPacketToSend::padding_size, kMaxPaddingLength)),
Pointee(Property(
@@ -677,8 +494,8 @@
start_time + kTimeDiff.ms())))));
}
-TEST_P(RtpSenderTest, KeepsTimestampsOnPayloadPadding) {
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+TEST_F(RtpSenderTest, KeepsTimestampsOnPayloadPadding) {
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
EnableRtx();
// Timestamps as set based on capture time in RtpSenderTest.
@@ -696,8 +513,7 @@
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);
+ packet_history_->PutRtpPacket(std::move(media_packet), start_time);
// Advance time before sending padding.
const TimeDelta kTimeDiff = TimeDelta::Millis(17);
@@ -705,8 +521,8 @@
// Timestamps on payload padding should be set to original.
EXPECT_THAT(
- rtp_sender()->GeneratePadding(/*target_size_bytes=*/100,
- /*media_has_been_sent=*/true),
+ 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,
@@ -717,7 +533,7 @@
// Test that the MID header extension is included on sent packets when
// configured.
-TEST_P(RtpSenderTest, MidIncludedOnSentPackets) {
+TEST_F(RtpSenderTest, MidIncludedOnSentPackets) {
const char kMid[] = "mid";
EnableMidSending(kMid);
@@ -730,7 +546,7 @@
SendGenericPacket();
}
-TEST_P(RtpSenderTest, RidIncludedOnSentPackets) {
+TEST_F(RtpSenderTest, RidIncludedOnSentPackets) {
const char kRid[] = "f";
EnableRidSending(kRid);
@@ -740,7 +556,7 @@
SendGenericPacket();
}
-TEST_P(RtpSenderTest, RidIncludedOnRtxSentPackets) {
+TEST_F(RtpSenderTest, RidIncludedOnRtxSentPackets) {
const char kRid[] = "f";
EnableRtx();
EnableRidSending(kRid);
@@ -751,8 +567,8 @@
Property(&RtpPacketToSend::HasExtension<RepairedRtpStreamId>,
false))))))
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packets[0]), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packets[0]),
+ clock_->TimeInMilliseconds());
});
SendGenericPacket();
@@ -761,10 +577,10 @@
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::GetExtension<RepairedRtpStreamId>, kRid),
Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
- rtp_sender()->ReSendPacket(kSeqNum);
+ rtp_sender_->ReSendPacket(kSeqNum);
}
-TEST_P(RtpSenderTest, MidAndRidNotIncludedOnSentPacketsAfterAck) {
+TEST_F(RtpSenderTest, MidAndRidNotIncludedOnSentPacketsAfterAck) {
const char kMid[] = "mid";
const char kRid[] = "f";
@@ -778,7 +594,7 @@
Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid))))));
auto first_built_packet = SendGenericPacket();
- rtp_sender()->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
+ rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
// The second packet should include neither since an ack was received.
EXPECT_CALL(
@@ -789,8 +605,8 @@
SendGenericPacket();
}
-TEST_P(RtpSenderTest, MidAndRidAlwaysIncludedOnSentPacketsWhenConfigured) {
- SetUpRtpSender(true, false, /*always_send_mid_and_rid=*/true, nullptr);
+TEST_F(RtpSenderTest, MidAndRidAlwaysIncludedOnSentPacketsWhenConfigured) {
+ SetUpRtpSender(false, /*always_send_mid_and_rid=*/true, nullptr);
const char kMid[] = "mid";
const char kRid[] = "f";
EnableMidSending(kMid);
@@ -805,7 +621,7 @@
Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid))))))
.Times(2);
auto first_built_packet = SendGenericPacket();
- rtp_sender()->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
+ rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
SendGenericPacket();
}
@@ -813,7 +629,7 @@
// 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(RtpSenderTest, MidAndRidIncludedOnFirstRtxPacket) {
+TEST_F(RtpSenderTest, MidAndRidIncludedOnFirstRtxPacket) {
const char kMid[] = "mid";
const char kRid[] = "f";
@@ -824,14 +640,14 @@
// 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());
+ 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) {
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packets[0]), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packets[0]),
+ clock_->TimeInMilliseconds());
});
auto second_built_packet = SendGenericPacket();
@@ -841,13 +657,13 @@
Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
Property(&RtpPacketToSend::GetExtension<RepairedRtpStreamId>,
kRid))))));
- rtp_sender()->ReSendPacket(second_built_packet->SequenceNumber());
+ 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_P(RtpSenderTest, MidAndRidNotIncludedOnRtxPacketsAfterAck) {
+TEST_F(RtpSenderTest, MidAndRidNotIncludedOnRtxPacketsAfterAck) {
const char kMid[] = "mid";
const char kRid[] = "f";
@@ -858,28 +674,28 @@
// This first packet will include both MID and RID.
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packets[0]), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packets[0]),
+ clock_->TimeInMilliseconds());
});
auto first_built_packet = SendGenericPacket();
- rtp_sender()->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
+ rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
// The second packet will include neither since an ack was received.
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packets[0]), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packets[0]),
+ clock_->TimeInMilliseconds());
});
auto second_built_packet = SendGenericPacket();
// 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());
- rtp_sender_context_->packet_history_.MarkPacketAsSent(
+ rtp_sender_->OnReceivedAckOnRtxSsrc(packets[0]->SequenceNumber());
+ packet_history_->MarkPacketAsSent(
*packets[0]->retransmitted_sequence_number());
});
- rtp_sender()->ReSendPacket(second_built_packet->SequenceNumber());
+ rtp_sender_->ReSendPacket(second_built_packet->SequenceNumber());
// The second and third RTX packets should not include MID nor RRID.
EXPECT_CALL(mock_paced_sender_,
@@ -888,12 +704,12 @@
Property(&RtpPacketToSend::HasExtension<RepairedRtpStreamId>,
false))))))
.Times(2);
- rtp_sender()->ReSendPacket(first_built_packet->SequenceNumber());
- rtp_sender()->ReSendPacket(second_built_packet->SequenceNumber());
+ rtp_sender_->ReSendPacket(first_built_packet->SequenceNumber());
+ rtp_sender_->ReSendPacket(second_built_packet->SequenceNumber());
}
-TEST_P(RtpSenderTest, MidAndRidAlwaysIncludedOnRtxPacketsWhenConfigured) {
- SetUpRtpSender(true, false, /*always_send_mid_and_rid=*/true, nullptr);
+TEST_F(RtpSenderTest, MidAndRidAlwaysIncludedOnRtxPacketsWhenConfigured) {
+ SetUpRtpSender(false, /*always_send_mid_and_rid=*/true, nullptr);
const char kMid[] = "mid";
const char kRid[] = "f";
EnableRtx();
@@ -909,11 +725,11 @@
.Times(2)
.WillRepeatedly(
[&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packets[0]), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packets[0]),
+ clock_->TimeInMilliseconds());
});
auto media_packet1 = SendGenericPacket();
- rtp_sender()->OnReceivedAckOnSsrc(media_packet1->SequenceNumber());
+ rtp_sender_->OnReceivedAckOnSsrc(media_packet1->SequenceNumber());
auto media_packet2 = SendGenericPacket();
// Send three RTX packets with different combinations of orders w.r.t. the
@@ -928,28 +744,28 @@
.Times(3)
.WillRepeatedly(
[&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
- rtp_sender()->OnReceivedAckOnRtxSsrc(packets[0]->SequenceNumber());
- rtp_sender_context_->packet_history_.MarkPacketAsSent(
+ 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());
+ 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_P(RtpSenderTest, MidAndRidNotIncludedOnSentPacketsAfterRtpStateRestored) {
+TEST_F(RtpSenderTest, MidAndRidNotIncludedOnSentPacketsAfterRtpStateRestored) {
const char kMid[] = "mid";
const char kRid[] = "f";
EnableMidSending(kMid);
EnableRidSending(kRid);
- RtpState state = rtp_sender()->GetRtpState();
+ RtpState state = rtp_sender_->GetRtpState();
EXPECT_FALSE(state.ssrc_has_acked);
state.ssrc_has_acked = true;
- rtp_sender()->SetRtpState(state);
+ rtp_sender_->SetRtpState(state);
EXPECT_CALL(
mock_paced_sender_,
@@ -962,7 +778,7 @@
// 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(RtpSenderTest, MidAndRridNotIncludedOnRtxPacketsAfterRtpStateRestored) {
+TEST_F(RtpSenderTest, MidAndRridNotIncludedOnRtxPacketsAfterRtpStateRestored) {
const char kMid[] = "mid";
const char kRid[] = "f";
@@ -970,15 +786,15 @@
EnableMidSending(kMid);
EnableRidSending(kRid);
- RtpState rtx_state = rtp_sender()->GetRtxRtpState();
+ RtpState rtx_state = rtp_sender_->GetRtxRtpState();
EXPECT_FALSE(rtx_state.ssrc_has_acked);
rtx_state.ssrc_has_acked = true;
- rtp_sender()->SetRtxRtpState(rtx_state);
+ rtp_sender_->SetRtxRtpState(rtx_state);
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packets[0]), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packets[0]),
+ clock_->TimeInMilliseconds());
});
auto built_packet = SendGenericPacket();
@@ -987,26 +803,26 @@
EnqueuePackets(ElementsAre(Pointee(AllOf(
Property(&RtpPacketToSend::HasExtension<RtpMid>, false),
Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
- ASSERT_LT(0, rtp_sender()->ReSendPacket(built_packet->SequenceNumber()));
+ ASSERT_LT(0, rtp_sender_->ReSendPacket(built_packet->SequenceNumber()));
}
-TEST_P(RtpSenderTest, RespectsNackBitrateLimit) {
+TEST_F(RtpSenderTest, RespectsNackBitrateLimit) {
const int32_t kPacketSize = 1400;
const int32_t kNumPackets = 30;
retransmission_rate_limiter_.SetMaxRate(kPacketSize * kNumPackets * 8);
- rtp_sender_context_->packet_history_.SetStorePacketsStatus(
+ packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, kNumPackets);
- const uint16_t kStartSequenceNumber = rtp_sender()->SequenceNumber();
+ 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));
EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
.WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packets[0]), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packets[0]),
+ clock_->TimeInMilliseconds());
});
SendPacket(clock_->TimeInMilliseconds(), kPacketSize);
}
@@ -1023,113 +839,96 @@
.WillRepeatedly(
[&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
for (const auto& packet : packets) {
- rtp_sender_context_->packet_history_.MarkPacketAsSent(
+ packet_history_->MarkPacketAsSent(
*packet->retransmitted_sequence_number());
}
});
- rtp_sender()->OnReceivedNack(sequence_numbers, 0);
+ 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);
+ rtp_sender_->OnReceivedNack(sequence_numbers, 0);
}
-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_);
+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);
+ 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);
+ 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_);
+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);
+ EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
- rtp_sender()->RegisterRtpHeaderExtension(TransmissionOffset::kUri,
- kTransmissionTimeOffsetExtensionId);
+ 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);
+ 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_);
+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);
+ EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
- rtp_sender()->RegisterRtpHeaderExtension(RtpMid::kUri, kMidExtensionId);
- rtp_sender()->RegisterRtpHeaderExtension(RtpStreamId::kUri, kRidExtensionId);
+ 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);
+ 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);
+ 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_);
+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);
+ 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);
+ 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);
+ EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
}
-TEST_P(RtpSenderTest, SendPacketHandlesRetransmissionHistory) {
- rtp_sender_context_->packet_history_.SetStorePacketsStatus(
+TEST_F(RtpSenderTest, SendPacketHandlesRetransmissionHistory) {
+ packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
// Ignore calls to EnqueuePackets() for this test.
@@ -1140,27 +939,27 @@
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());
+ 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));
+ 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));
+ 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);
+ 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));
+ EXPECT_THAT(rtp_sender_->ReSendPacket(media_sequence_number), Gt(0));
}
-TEST_P(RtpSenderTest, MarksRetransmittedPackets) {
- rtp_sender_context_->packet_history_.SetStorePacketsStatus(
+TEST_F(RtpSenderTest, MarksRetransmittedPackets) {
+ packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
// Build a media packet and put in the packet history.
@@ -1168,8 +967,8 @@
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());
+ packet_history_->PutRtpPacket(std::move(packet),
+ clock_->TimeInMilliseconds());
// Expect a retransmission packet marked with which packet it is a
// retransmit of.
@@ -1180,19 +979,19 @@
RtpPacketMediaType::kRetransmission)),
Pointee(Property(&RtpPacketToSend::retransmitted_sequence_number,
Eq(media_sequence_number)))))));
- EXPECT_THAT(rtp_sender()->ReSendPacket(media_sequence_number), Gt(0));
+ EXPECT_THAT(rtp_sender_->ReSendPacket(media_sequence_number), Gt(0));
}
-TEST_P(RtpSenderTest, GeneratedPaddingHasBweExtensions) {
+TEST_F(RtpSenderTest, GeneratedPaddingHasBweExtensions) {
// Min requested size in order to use RTX payload.
const size_t kMinPaddingSize = 50;
EnableRtx();
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransmissionOffset::kUri, kTransmissionTimeOffsetExtensionId));
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
// Put a packet in the history, in order to facilitate payload padding.
@@ -1201,12 +1000,12 @@
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kMinPaddingSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packet), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packet),
+ clock_->TimeInMilliseconds());
// 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);
+ 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);
@@ -1216,7 +1015,7 @@
EXPECT_GT(plain_padding->padding_size(), 0u);
// Generate a payload padding packets, check that extensions are registered.
- generated_packets = rtp_sender()->GeneratePadding(kMinPaddingSize, true);
+ 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);
@@ -1226,16 +1025,16 @@
EXPECT_GT(payload_padding->payload_size(), 0u);
}
-TEST_P(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
+TEST_F(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(
+ rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+ rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
+ packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
const size_t kPayloadPacketSize = kMinPaddingSize;
@@ -1244,13 +1043,13 @@
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packet), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packet),
+ clock_->TimeInMilliseconds());
// 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);
+ 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);
@@ -1263,7 +1062,7 @@
size_t padding_bytes_generated = 0;
generated_packets =
- rtp_sender()->GeneratePadding(kPaddingBytesRequested, true);
+ rtp_sender_->GeneratePadding(kPaddingBytesRequested, true);
EXPECT_EQ(generated_packets.size(), 1u);
for (auto& packet : generated_packets) {
EXPECT_EQ(packet->packet_type(), RtpPacketMediaType::kPadding);
@@ -1276,17 +1075,17 @@
EXPECT_EQ(padding_bytes_generated, kMaxPaddingSize);
}
-TEST_P(RtpSenderTest, LimitsPayloadPaddingSize) {
+TEST_F(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(
+ SetUpRtpSender(false, false, nullptr);
+ rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+ rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
+ packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
// Send a dummy video packet so it ends up in the packet history.
@@ -1296,8 +1095,8 @@
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packet), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packet),
+ clock_->TimeInMilliseconds());
// Smallest target size that will result in the sent packet being returned as
// padding.
@@ -1307,26 +1106,26 @@
// Generated padding has large enough budget that the video packet should be
// retransmitted as padding.
EXPECT_THAT(
- rtp_sender()->GeneratePadding(kMinTargerSizeForPayload, true),
+ 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),
+ 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(
+TEST_F(RtpSenderTest, GeneratePaddingCreatesPurePaddingWithoutRtx) {
+ packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 1);
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransmissionOffset::kUri, kTransmissionTimeOffsetExtensionId));
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
AbsoluteSendTime::kUri, kAbsoluteSendTimeExtensionId));
- ASSERT_TRUE(rtp_sender()->RegisterRtpHeaderExtension(
+ ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, kTransportSequenceNumberExtensionId));
const size_t kPayloadPacketSize = 1234;
@@ -1337,8 +1136,8 @@
packet->set_allow_retransmission(true);
packet->SetPayloadSize(kPayloadPacketSize);
packet->set_packet_type(RtpPacketMediaType::kVideo);
- rtp_sender_context_->packet_history_.PutRtpPacket(
- std::move(packet), clock_->TimeInMilliseconds());
+ packet_history_->PutRtpPacket(std::move(packet),
+ clock_->TimeInMilliseconds());
// Payload padding not available without RTX, only generate plain padding on
// the media SSRC.
@@ -1350,7 +1149,7 @@
(kPaddingBytesRequested + kMaxPaddingSize - 1) / kMaxPaddingSize;
size_t padding_bytes_generated = 0;
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
- rtp_sender()->GeneratePadding(kPaddingBytesRequested, true);
+ rtp_sender_->GeneratePadding(kPaddingBytesRequested, true);
EXPECT_EQ(padding_packets.size(), kExpectedNumPaddingPackets);
for (auto& packet : padding_packets) {
EXPECT_EQ(packet->packet_type(), RtpPacketMediaType::kPadding);
@@ -1367,7 +1166,7 @@
kExpectedNumPaddingPackets * kMaxPaddingSize);
}
-TEST_P(RtpSenderTest, SupportsPadding) {
+TEST_F(RtpSenderTest, SupportsPadding) {
bool kSendingMediaStats[] = {true, false};
bool kEnableRedundantPayloads[] = {true, false};
absl::string_view kBweExtensionUris[] = {
@@ -1376,35 +1175,35 @@
const int kExtensionsId = 7;
for (bool sending_media : kSendingMediaStats) {
- rtp_sender()->SetSendingMediaStatus(sending_media);
+ 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);
+ rtp_sender_->SetRtxStatus(rtx_mode);
for (auto extension_uri : kBweExtensionUris) {
- EXPECT_FALSE(rtp_sender()->SupportsPadding());
- rtp_sender()->RegisterRtpHeaderExtension(extension_uri, kExtensionsId);
+ EXPECT_FALSE(rtp_sender_->SupportsPadding());
+ rtp_sender_->RegisterRtpHeaderExtension(extension_uri, kExtensionsId);
if (!sending_media) {
- EXPECT_FALSE(rtp_sender()->SupportsPadding());
+ EXPECT_FALSE(rtp_sender_->SupportsPadding());
} else {
- EXPECT_TRUE(rtp_sender()->SupportsPadding());
+ EXPECT_TRUE(rtp_sender_->SupportsPadding());
if (redundant_payloads) {
- EXPECT_TRUE(rtp_sender()->SupportsRtxPayloadPadding());
+ EXPECT_TRUE(rtp_sender_->SupportsRtxPayloadPadding());
} else {
- EXPECT_FALSE(rtp_sender()->SupportsRtxPayloadPadding());
+ EXPECT_FALSE(rtp_sender_->SupportsRtxPayloadPadding());
}
}
- rtp_sender()->DeregisterRtpHeaderExtension(extension_uri);
- EXPECT_FALSE(rtp_sender()->SupportsPadding());
+ rtp_sender_->DeregisterRtpHeaderExtension(extension_uri);
+ EXPECT_FALSE(rtp_sender_->SupportsPadding());
}
}
}
}
-TEST_P(RtpSenderTest, SetsCaptureTimeOnRtxRetransmissions) {
+TEST_F(RtpSenderTest, SetsCaptureTimeOnRtxRetransmissions) {
EnableRtx();
// Put a packet in the packet history, with current time as capture time.
@@ -1413,8 +1212,7 @@
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);
+ packet_history_->PutRtpPacket(std::move(packet), start_time_ms);
// Advance time, request an RTX retransmission. Capture timestamp should be
// preserved.
@@ -1423,10 +1221,10 @@
EXPECT_CALL(mock_paced_sender_,
EnqueuePackets(ElementsAre(Pointee(Property(
&RtpPacketToSend::capture_time_ms, start_time_ms)))));
- EXPECT_GT(rtp_sender()->ReSendPacket(kSeqNum), 0);
+ EXPECT_GT(rtp_sender_->ReSendPacket(kSeqNum), 0);
}
-TEST_P(RtpSenderTestWithoutPacer, ClearHistoryOnSequenceNumberCange) {
+TEST_F(RtpSenderTest, ClearHistoryOnSequenceNumberCange) {
EnableRtx();
// Put a packet in the packet history.
@@ -1434,21 +1232,21 @@
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);
+ packet_history_->PutRtpPacket(std::move(packet), now_ms);
- EXPECT_TRUE(rtp_sender_context_->packet_history_.GetPacketState(kSeqNum));
+ EXPECT_TRUE(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));
+ rtp_sender_->SetSequenceNumber(rtp_sender_->SequenceNumber() - 1);
+ EXPECT_FALSE(packet_history_->GetPacketState(kSeqNum));
}
-TEST_P(RtpSenderTest, IgnoresNackAfterDisablingMedia) {
+TEST_F(RtpSenderTest, IgnoresNackAfterDisablingMedia) {
const int64_t kRtt = 10;
EnableRtx();
- rtp_sender_context_->packet_history_.SetRtt(kRtt);
+ packet_history_->SetRtt(kRtt);
// Put a packet in the history.
const int64_t start_time_ms = clock_->TimeInMilliseconds();
@@ -1456,24 +1254,23 @@
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);
+ 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);
+ rtp_sender_->SetSendingMediaStatus(false);
time_controller_.AdvanceTime(TimeDelta::Millis(kRtt));
- EXPECT_LT(rtp_sender()->ReSendPacket(kSeqNum), 0);
+ EXPECT_LT(rtp_sender_->ReSendPacket(kSeqNum), 0);
}
-TEST_P(RtpSenderTest, DoesntFecProtectRetransmissions) {
+TEST_F(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(
+ rtp_sender_->SetSendingMediaStatus(true);
+ rtp_sender_->SetRtxStatus(kRtxOff);
+ packet_history_->SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
- rtp_sender_context_->packet_history_.SetRtt(kRtt);
+ packet_history_->SetRtt(kRtt);
// Put a fec protected packet in the history.
const int64_t start_time_ms = clock_->TimeInMilliseconds();
@@ -1482,8 +1279,7 @@
/*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);
+ packet_history_->PutRtpPacket(std::move(packet), start_time_ms);
// Re-send packet, the retransmitted packet should not have the FEC protection
// flag set.
@@ -1492,14 +1288,14 @@
Property(&RtpPacketToSend::fec_protect_packet, false)))));
time_controller_.AdvanceTime(TimeDelta::Millis(kRtt));
- EXPECT_GT(rtp_sender()->ReSendPacket(kSeqNum), 0);
+ EXPECT_GT(rtp_sender_->ReSendPacket(kSeqNum), 0);
}
-TEST_P(RtpSenderTest, MarksPacketsWithKeyframeStatus) {
+TEST_F(RtpSenderTest, MarksPacketsWithKeyframeStatus) {
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = clock_;
- video_config.rtp_sender = rtp_sender();
+ video_config.rtp_sender = rtp_sender_.get();
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
@@ -1542,14 +1338,4 @@
}
}
-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
