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webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
Niels Möller 5fe9510efb Move ownership of RTPSenderVideo one more level up, to RtpVideoSender 
The idea is to let the RtpRtcp and RTPSender classes be responsible for
media-agnostic RTP transport, and move out the media-specific processing,
such as packetization and media-specific headers.

Bug: webrtc:7135
Change-Id: Ib0ce45bf06713b3eb6c06acd91c5168856874e4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123187
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26954}
2019-03-04 16:57:49 +00:00

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include <vector>
#include "absl/memory/memory.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_codec_constants.h"
#include "api/video/video_timing.h"
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.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.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/rate_limiter.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_transport.h"
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kAbsoluteSendTimeExtensionId = 1,
kAudioLevelExtensionId,
kGenericDescriptorId00,
kGenericDescriptorId01,
kMidExtensionId,
kRepairedRidExtensionId,
kRidExtensionId,
kTransmissionTimeOffsetExtensionId,
kTransportSequenceNumberExtensionId,
kVideoRotationExtensionId,
kVideoTimingExtensionId,
};
const int kPayload = 100;
const int kRtxPayload = 98;
const uint32_t kTimestamp = 10;
const uint16_t kSeqNum = 33;
const uint32_t kSsrc = 725242;
const uint16_t kTransportSequenceNumber = 0xaabbu;
const uint64_t kStartTime = 123456789;
const size_t kMaxPaddingSize = 224u;
const uint8_t kPayloadData[] = {47, 11, 32, 93, 89};
const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
const char kNoRid[] = "";
const char kNoMid[] = "";
using ::testing::_;
using ::testing::ElementsAre;
using ::testing::ElementsAreArray;
using ::testing::Invoke;
using ::testing::SizeIs;
uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;
}
class LoopbackTransportTest : public webrtc::Transport {
public:
LoopbackTransportTest() : total_bytes_sent_(0) {
receivers_extensions_.Register(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
receivers_extensions_.Register(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId);
receivers_extensions_.Register(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
receivers_extensions_.Register(kRtpExtensionVideoRotation,
kVideoRotationExtensionId);
receivers_extensions_.Register(kRtpExtensionAudioLevel,
kAudioLevelExtensionId);
receivers_extensions_.Register(kRtpExtensionVideoTiming,
kVideoTimingExtensionId);
receivers_extensions_.Register(kRtpExtensionMid, kMidExtensionId);
receivers_extensions_.Register(kRtpExtensionGenericFrameDescriptor00,
kGenericDescriptorId00);
receivers_extensions_.Register(kRtpExtensionGenericFrameDescriptor01,
kGenericDescriptorId01);
receivers_extensions_.Register(kRtpExtensionRtpStreamId, kRidExtensionId);
receivers_extensions_.Register(kRtpExtensionRepairedRtpStreamId,
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();
}
} // namespace
class MockRtpPacketSender : public RtpPacketSender {
public:
MockRtpPacketSender() {}
virtual ~MockRtpPacketSender() {}
MOCK_METHOD6(InsertPacket,
void(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission));
};
class MockTransportSequenceNumberAllocator
: public TransportSequenceNumberAllocator {
public:
MOCK_METHOD0(AllocateSequenceNumber, uint16_t());
};
class MockSendSideDelayObserver : public SendSideDelayObserver {
public:
MOCK_METHOD3(SendSideDelayUpdated, void(int, int, uint32_t));
};
class MockSendPacketObserver : public SendPacketObserver {
public:
MOCK_METHOD3(OnSendPacket, void(uint16_t, int64_t, uint32_t));
};
class MockTransportFeedbackObserver : public TransportFeedbackObserver {
public:
MOCK_METHOD4(AddPacket,
void(uint32_t, uint16_t, size_t, const PacedPacketInfo&));
MOCK_METHOD1(OnTransportFeedback, void(const rtcp::TransportFeedback&));
MOCK_CONST_METHOD0(GetTransportFeedbackVector, std::vector<PacketFeedback>());
};
class MockOverheadObserver : public OverheadObserver {
public:
MOCK_METHOD1(OnOverheadChanged, void(size_t overhead_bytes_per_packet));
};
class RtpSenderTest : public ::testing::TestWithParam<bool> {
protected:
RtpSenderTest()
: fake_clock_(kStartTime),
mock_rtc_event_log_(),
mock_paced_sender_(),
retransmission_rate_limiter_(&fake_clock_, 1000),
rtp_sender_(),
transport_(),
kMarkerBit(true),
field_trials_(GetParam() ? "WebRTC-SendSideBwe-WithOverhead/Enabled/"
: "") {}
void SetUp() override { SetUpRtpSender(true, false); }
void SetUpRtpSender(bool pacer, bool populate_network2) {
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
absl::nullopt, &seq_num_allocator_, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, populate_network2, nullptr,
false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
}
SimulatedClock fake_clock_;
testing::NiceMock<MockRtcEventLog> mock_rtc_event_log_;
MockRtpPacketSender mock_paced_sender_;
testing::StrictMock<MockTransportSequenceNumberAllocator> seq_num_allocator_;
testing::StrictMock<MockSendPacketObserver> send_packet_observer_;
testing::StrictMock<MockTransportFeedbackObserver> feedback_observer_;
RateLimiter retransmission_rate_limiter_;
std::unique_ptr<RTPSender> rtp_sender_;
LoopbackTransportTest transport_;
const bool kMarkerBit;
test::ScopedFieldTrials field_trials_;
std::unique_ptr<RtpPacketToSend> BuildRtpPacket(int payload_type,
bool marker_bit,
uint32_t timestamp,
int64_t capture_time_ms) {
auto packet = rtp_sender_->AllocatePacket();
packet->SetPayloadType(payload_type);
packet->SetMarker(marker_bit);
packet->SetTimestamp(timestamp);
packet->set_capture_time_ms(capture_time_ms);
EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
return packet;
}
void SendPacket(int64_t capture_time_ms, int payload_length) {
uint32_t timestamp = capture_time_ms * 90;
auto packet =
BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
packet->AllocatePayload(payload_length);
// Packet should be stored in a send bucket.
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
}
void SendGenericPacket() {
const int64_t kCaptureTimeMs = fake_clock_.TimeInMilliseconds();
SendPacket(kCaptureTimeMs, sizeof(kPayloadData));
}
};
// 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); }
};
TEST_P(RtpSenderTestWithoutPacer, AllocatePacketSetCsrc) {
// Configure rtp_sender with csrc.
std::vector<uint32_t> csrcs;
csrcs.push_back(0x23456789);
rtp_sender_->SetCsrcs(csrcs);
auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet);
EXPECT_EQ(rtp_sender_->SSRC(), packet->Ssrc());
EXPECT_EQ(csrcs, packet->Csrcs());
}
TEST_P(RtpSenderTestWithoutPacer, AllocatePacketReserveExtensions) {
// Configure rtp_sender with extensions.
ASSERT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
ASSERT_EQ(
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
ASSERT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
kAudioLevelExtensionId));
ASSERT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
ASSERT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoRotation, kVideoRotationExtensionId));
auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet);
// Preallocate BWE extensions RtpSender set itself.
EXPECT_TRUE(packet->HasExtension<TransmissionOffset>());
EXPECT_TRUE(packet->HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(packet->HasExtension<TransportSequenceNumber>());
// Do not allocate media specific extensions.
EXPECT_FALSE(packet->HasExtension<AudioLevel>());
EXPECT_FALSE(packet->HasExtension<VideoOrientation>());
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberAdvanceSequenceNumber) {
auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet);
const uint16_t sequence_number = rtp_sender_->SequenceNumber();
EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
EXPECT_EQ(sequence_number, packet->SequenceNumber());
EXPECT_EQ(sequence_number + 1, rtp_sender_->SequenceNumber());
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberFailsOnNotSending) {
auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet);
rtp_sender_->SetSendingMediaStatus(false);
EXPECT_FALSE(rtp_sender_->AssignSequenceNumber(packet.get()));
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPaddingOnVideo) {
constexpr size_t kPaddingSize = 100;
auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet);
ASSERT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
packet->SetMarker(false);
ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
// Packet without marker bit doesn't allow padding on video stream.
EXPECT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
packet->SetMarker(true);
ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
// Packet with marker bit allows send padding.
EXPECT_TRUE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
}
TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
MockTransport transport;
const bool kEnableAudio = true;
rtp_sender_.reset(new RTPSender(
kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_,
absl::nullopt, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, nullptr, false, nullptr, false,
false, FieldTrialBasedConfig()));
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
std::unique_ptr<RtpPacketToSend> audio_packet = rtp_sender_->AllocatePacket();
// Padding on audio stream allowed regardless of marker in the last packet.
audio_packet->SetMarker(false);
audio_packet->SetPayloadType(kPayload);
rtp_sender_->AssignSequenceNumber(audio_packet.get());
const size_t kPaddingSize = 59;
EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kPaddingSize,
rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
// Requested padding size is too small, will send a larger one.
const size_t kMinPaddingSize = 50;
EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kMinPaddingSize, rtp_sender_->TimeToSendPadding(kMinPaddingSize - 5,
PacedPacketInfo()));
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
constexpr size_t kPaddingSize = 100;
auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet);
packet->SetMarker(true);
packet->SetTimestamp(kTimestamp);
ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
ASSERT_TRUE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
ASSERT_EQ(1u, transport_.sent_packets_.size());
// Verify padding packet timestamp.
EXPECT_EQ(kTimestamp, transport_.last_sent_packet().Timestamp());
}
TEST_P(RtpSenderTestWithoutPacer,
TransportFeedbackObserverGetsCorrectByteCount) {
constexpr int kRtpOverheadBytesPerPacket = 12 + 8;
testing::NiceMock<MockOverheadObserver> mock_overhead_observer;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
&seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
&mock_rtc_event_log_, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer,
false, nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
.WillOnce(testing::Return(kTransportSequenceNumber));
const size_t expected_bytes =
GetParam() ? sizeof(kPayloadData) + kRtpOverheadBytesPerPacket
: sizeof(kPayloadData);
EXPECT_CALL(feedback_observer_,
AddPacket(rtp_sender_->SSRC(), kTransportSequenceNumber,
expected_bytes, PacedPacketInfo()))
.Times(1);
EXPECT_CALL(mock_overhead_observer,
OnOverheadChanged(kRtpOverheadBytesPerPacket))
.Times(1);
SendGenericPacket();
}
TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
&seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, false, nullptr,
false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
.WillOnce(testing::Return(kTransportSequenceNumber));
EXPECT_CALL(send_packet_observer_,
OnSendPacket(kTransportSequenceNumber, _, _))
.Times(1);
EXPECT_CALL(feedback_observer_,
AddPacket(rtp_sender_->SSRC(), kTransportSequenceNumber, _,
PacedPacketInfo()))
.Times(1);
SendGenericPacket();
const auto& packet = transport_.last_sent_packet();
uint16_t transport_seq_no;
ASSERT_TRUE(packet.GetExtension<TransportSequenceNumber>(&transport_seq_no));
EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
EXPECT_EQ(transport_.last_options_.packet_id, transport_seq_no);
EXPECT_TRUE(transport_.last_options_.included_in_allocation);
}
TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
&seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, false, nullptr,
false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
SendGenericPacket();
EXPECT_FALSE(transport_.last_options_.is_retransmit);
}
TEST_P(RtpSenderTestWithoutPacer,
SetsIncludedInFeedbackWhenTransportSequenceNumberExtensionIsRegistered) {
SetUpRtpSender(false, false);
rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
.WillOnce(testing::Return(kTransportSequenceNumber));
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
SendGenericPacket();
EXPECT_TRUE(transport_.last_options_.included_in_feedback);
}
TEST_P(
RtpSenderTestWithoutPacer,
SetsIncludedInAllocationWhenTransportSequenceNumberExtensionIsRegistered) {
SetUpRtpSender(false, false);
rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
.WillOnce(testing::Return(kTransportSequenceNumber));
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
SendGenericPacket();
EXPECT_TRUE(transport_.last_options_.included_in_allocation);
}
TEST_P(RtpSenderTestWithoutPacer,
SetsIncludedInAllocationWhenForcedAsPartOfAllocation) {
SetUpRtpSender(false, false);
rtp_sender_->SetAsPartOfAllocation(true);
SendGenericPacket();
EXPECT_FALSE(transport_.last_options_.included_in_feedback);
EXPECT_TRUE(transport_.last_options_.included_in_allocation);
}
TEST_P(RtpSenderTestWithoutPacer, DoesnSetIncludedInAllocationByDefault) {
SetUpRtpSender(false, false);
SendGenericPacket();
EXPECT_FALSE(transport_.last_options_.included_in_feedback);
EXPECT_FALSE(transport_.last_options_.included_in_allocation);
}
TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
testing::StrictMock<MockSendSideDelayObserver> send_side_delay_observer_;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
nullptr, nullptr, nullptr, &send_side_delay_observer_,
&mock_rtc_event_log_, nullptr, nullptr, nullptr, false,
nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
PlayoutDelayOracle playout_delay_oracle;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
&playout_delay_oracle, nullptr, false,
FieldTrialBasedConfig());
const uint8_t kPayloadType = 127;
const char payload_name[] = "GENERIC";
rtp_sender_video.RegisterPayloadType(kPayloadType, payload_name);
const uint32_t kCaptureTimeMsToRtpTimestamp = 90; // 90 kHz clock
RTPVideoHeader video_header;
// Send packet with 10 ms send-side delay. The average and max should be 10
// ms.
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(10, 10, kSsrc))
.Times(1);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
fake_clock_.AdvanceTimeMilliseconds(10);
EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
// Send another packet with 20 ms delay. The average
// and max should be 15 and 20 ms respectively.
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(15, 20, kSsrc))
.Times(1);
fake_clock_.AdvanceTimeMilliseconds(10);
EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
// Send another packet at the same time, which replaces the last packet.
// Since this packet has 0 ms delay, the average is now 5 ms and max is 10 ms.
// TODO(terelius): Is is not clear that this is the right behavior.
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(5, 10, kSsrc))
.Times(1);
capture_time_ms = fake_clock_.TimeInMilliseconds();
EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
// Send a packet 1 second later. The earlier packets should have timed
// out, so both max and average should be the delay of this packet.
fake_clock_.AdvanceTimeMilliseconds(1000);
capture_time_ms = fake_clock_.TimeInMilliseconds();
fake_clock_.AdvanceTimeMilliseconds(1);
EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(1, 1, kSsrc))
.Times(1);
EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kPayloadType,
capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
}
TEST_P(RtpSenderTestWithoutPacer, OnSendPacketUpdated) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
.WillOnce(testing::Return(kTransportSequenceNumber));
EXPECT_CALL(send_packet_observer_,
OnSendPacket(kTransportSequenceNumber, _, _))
.Times(1);
SendGenericPacket();
}
TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
&seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, kSsrc, kSeqNum, _, _, _));
EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
.WillOnce(testing::Return(kTransportSequenceNumber));
EXPECT_CALL(send_packet_observer_,
OnSendPacket(kTransportSequenceNumber, _, _))
.Times(1);
EXPECT_CALL(feedback_observer_,
AddPacket(rtp_sender_->SSRC(), kTransportSequenceNumber, _,
PacedPacketInfo()))
.Times(1);
SendGenericPacket();
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum,
fake_clock_.TimeInMilliseconds(), false,
PacedPacketInfo());
const auto& packet = transport_.last_sent_packet();
uint16_t transport_seq_no;
EXPECT_TRUE(packet.GetExtension<TransportSequenceNumber>(&transport_seq_no));
EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
EXPECT_EQ(transport_.last_options_.packet_id, transport_seq_no);
}
TEST_P(RtpSenderTest, WritesPacerExitToTimingExtension) {
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoTiming, kVideoTimingExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet = rtp_sender_->AllocatePacket();
packet->SetPayloadType(kPayload);
packet->SetMarker(true);
packet->SetTimestamp(kTimestamp);
packet->set_capture_time_ms(capture_time_ms);
const VideoSendTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};
packet->SetExtension<VideoTimingExtension>(kVideoTiming);
EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
size_t packet_size = packet->size();
const int kStoredTimeInMs = 100;
{
EXPECT_CALL(
mock_paced_sender_,
InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _));
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
}
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
VideoSendTiming video_timing;
EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
&video_timing));
EXPECT_EQ(kStoredTimeInMs, video_timing.pacer_exit_delta_ms);
}
TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtensionWithPacer) {
SetUpRtpSender(/*pacer=*/true, /*populate_network2=*/true);
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoTiming, kVideoTimingExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet = rtp_sender_->AllocatePacket();
packet->SetPayloadType(kPayload);
packet->SetMarker(true);
packet->SetTimestamp(kTimestamp);
packet->set_capture_time_ms(capture_time_ms);
const uint16_t kPacerExitMs = 1234u;
const VideoSendTiming kVideoTiming = {0u, 0u, 0u, kPacerExitMs, 0u, 0u, true};
packet->SetExtension<VideoTimingExtension>(kVideoTiming);
EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
size_t packet_size = packet->size();
const int kStoredTimeInMs = 100;
{
EXPECT_CALL(
mock_paced_sender_,
InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _));
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
}
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
VideoSendTiming video_timing;
EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
&video_timing));
EXPECT_EQ(kStoredTimeInMs, video_timing.network2_timestamp_delta_ms);
EXPECT_EQ(kPacerExitMs, video_timing.pacer_exit_delta_ms);
}
TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtensionWithoutPacer) {
SetUpRtpSender(/*pacer=*/false, /*populate_network2=*/true);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoTiming, kVideoTimingExtensionId));
auto packet = rtp_sender_->AllocatePacket();
packet->SetMarker(true);
packet->set_capture_time_ms(fake_clock_.TimeInMilliseconds());
const VideoSendTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};
packet->SetExtension<VideoTimingExtension>(kVideoTiming);
EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
const int kPropagateTimeMs = 10;
fake_clock_.AdvanceTimeMilliseconds(kPropagateTimeMs);
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_EQ(1, transport_.packets_sent());
absl::optional<VideoSendTiming> video_timing =
transport_.last_sent_packet().GetExtension<VideoTimingExtension>();
ASSERT_TRUE(video_timing);
EXPECT_EQ(kPropagateTimeMs, video_timing->network2_timestamp_delta_ms);
}
TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
kSsrc, kSeqNum, _, _, _));
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)));
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet =
BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, capture_time_ms);
size_t packet_size = packet->size();
// Packet should be stored in a send bucket.
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_EQ(0, transport_.packets_sent());
const int kStoredTimeInMs = 100;
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
PacedPacketInfo());
// Process send bucket. Packet should now be sent.
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
webrtc::RTPHeader rtp_header;
transport_.last_sent_packet().GetHeader(&rtp_header);
// Verify transmission time offset.
EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset);
uint64_t expected_send_time =
ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
}
TEST_P(RtpSenderTest, TrafficSmoothingRetransmits) {
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
kSsrc, kSeqNum, _, _, _));
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)));
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet =
BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, capture_time_ms);
size_t packet_size = packet->size();
// Packet should be stored in a send bucket.
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_EQ(0, transport_.packets_sent());
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
kSsrc, kSeqNum, _, _, _));
const int kStoredTimeInMs = 100;
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
EXPECT_EQ(static_cast<int>(packet_size), rtp_sender_->ReSendPacket(kSeqNum));
EXPECT_EQ(0, transport_.packets_sent());
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
PacedPacketInfo());
// Process send bucket. Packet should now be sent.
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
webrtc::RTPHeader rtp_header;
transport_.last_sent_packet().GetHeader(&rtp_header);
// Verify transmission time offset.
EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset);
uint64_t expected_send_time =
ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
}
// This test sends 1 regular video packet, then 4 padding packets, and then
// 1 more regular packet.
TEST_P(RtpSenderTest, SendPadding) {
// Make all (non-padding) packets go to send queue.
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
kSsrc, kSeqNum, _, _, _));
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(1 + 4 + 1);
uint16_t seq_num = kSeqNum;
uint32_t timestamp = kTimestamp;
rtp_sender_->SetStorePacketsStatus(true, 10);
size_t rtp_header_len = kRtpHeaderSize;
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
rtp_header_len += 4; // 4 bytes extension.
EXPECT_EQ(
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
rtp_header_len += 4; // 4 bytes extension.
rtp_header_len += 4; // 4 extra bytes common to all extension headers.
webrtc::RTPHeader rtp_header;
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet =
BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
const uint32_t media_packet_timestamp = timestamp;
size_t packet_size = packet->size();
// Packet should be stored in a send bucket.
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
int total_packets_sent = 0;
EXPECT_EQ(total_packets_sent, transport_.packets_sent());
const int kStoredTimeInMs = 100;
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_sender_->TimeToSendPacket(kSsrc, seq_num++, capture_time_ms, false,
PacedPacketInfo());
// Packet should now be sent. This test doesn't verify the regular video
// packet, since it is tested in another test.
EXPECT_EQ(++total_packets_sent, transport_.packets_sent());
timestamp += 90 * kStoredTimeInMs;
// Send padding 4 times, waiting 50 ms between each.
for (int i = 0; i < 4; ++i) {
const int kPaddingPeriodMs = 50;
const size_t kPaddingBytes = 100;
const size_t kMaxPaddingLength = 224; // Value taken from rtp_sender.cc.
// Padding will be forced to full packets.
EXPECT_EQ(kMaxPaddingLength,
rtp_sender_->TimeToSendPadding(kPaddingBytes, PacedPacketInfo()));
// Process send bucket. Padding should now be sent.
EXPECT_EQ(++total_packets_sent, transport_.packets_sent());
EXPECT_EQ(kMaxPaddingLength + rtp_header_len,
transport_.last_sent_packet().size());
transport_.last_sent_packet().GetHeader(&rtp_header);
EXPECT_EQ(kMaxPaddingLength, rtp_header.paddingLength);
// Verify sequence number and timestamp. The timestamp should be the same
// as the last media packet.
EXPECT_EQ(seq_num++, rtp_header.sequenceNumber);
EXPECT_EQ(media_packet_timestamp, rtp_header.timestamp);
// Verify transmission time offset.
int offset = timestamp - media_packet_timestamp;
EXPECT_EQ(offset, rtp_header.extension.transmissionTimeOffset);
uint64_t expected_send_time =
ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
fake_clock_.AdvanceTimeMilliseconds(kPaddingPeriodMs);
timestamp += 90 * kPaddingPeriodMs;
}
// Send a regular video packet again.
capture_time_ms = fake_clock_.TimeInMilliseconds();
packet = BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
packet_size = packet->size();
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
kSsrc, seq_num, _, _, _));
// Packet should be stored in a send bucket.
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
rtp_sender_->TimeToSendPacket(kSsrc, seq_num, capture_time_ms, false,
PacedPacketInfo());
// Process send bucket.
EXPECT_EQ(++total_packets_sent, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
transport_.last_sent_packet().GetHeader(&rtp_header);
// Verify sequence number and timestamp.
EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
EXPECT_EQ(timestamp, rtp_header.timestamp);
// Verify transmission time offset. This packet is sent without delay.
EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
uint64_t expected_send_time =
ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
}
TEST_P(RtpSenderTest, OnSendPacketUpdated) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_CALL(send_packet_observer_,
OnSendPacket(kTransportSequenceNumber, _, _))
.Times(1);
EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
.WillOnce(testing::Return(kTransportSequenceNumber));
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, kSsrc, kSeqNum, _, _, _))
.Times(1);
SendGenericPacket(); // Packet passed to pacer.
const bool kIsRetransmit = false;
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum,
fake_clock_.TimeInMilliseconds(), kIsRetransmit,
PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
}
TEST_P(RtpSenderTest, OnSendPacketNotUpdatedForRetransmits) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_CALL(send_packet_observer_, OnSendPacket(_, _, _)).Times(0);
EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
.WillOnce(testing::Return(kTransportSequenceNumber));
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, kSsrc, kSeqNum, _, _, _))
.Times(1);
SendGenericPacket(); // Packet passed to pacer.
const bool kIsRetransmit = true;
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum,
fake_clock_.TimeInMilliseconds(), kIsRetransmit,
PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_TRUE(transport_.last_options_.is_retransmit);
}
TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
nullptr, &send_packet_observer_, &retransmission_rate_limiter_, nullptr,
false, nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_CALL(send_packet_observer_, OnSendPacket(_, _, _)).Times(0);
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, kSsrc, kSeqNum, _, _, _))
.Times(1);
SendGenericPacket(); // Packet passed to pacer.
const bool kIsRetransmit = false;
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum,
fake_clock_.TimeInMilliseconds(), kIsRetransmit,
PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
}
TEST_P(RtpSenderTest, SendRedundantPayloads) {
MockTransport transport;
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport, &mock_paced_sender_, absl::nullopt,
nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
&retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
uint16_t seq_num = kSeqNum;
rtp_sender_->SetStorePacketsStatus(true, 10);
int32_t rtp_header_len = kRtpHeaderSize;
EXPECT_EQ(
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
rtp_header_len += 4; // 4 bytes extension.
rtp_header_len += 4; // 4 extra bytes common to all extension headers.
rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
rtp_sender_->SetRtxSsrc(1234);
const size_t kNumPayloadSizes = 10;
const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700,
750, 800, 850, 900, 950};
// Expect all packets go through the pacer.
EXPECT_CALL(mock_paced_sender_,
InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _))
.Times(kNumPayloadSizes);
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(kNumPayloadSizes);
// Send 10 packets of increasing size.
for (size_t i = 0; i < kNumPayloadSizes; ++i) {
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
EXPECT_CALL(transport, SendRtp(_, _, _)).WillOnce(testing::Return(true));
SendPacket(capture_time_ms, kPayloadSizes[i]);
rtp_sender_->TimeToSendPacket(kSsrc, seq_num++, capture_time_ms, false,
PacedPacketInfo());
fake_clock_.AdvanceTimeMilliseconds(33);
}
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(::testing::AtLeast(4));
// The amount of padding to send it too small to send a payload packet.
EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kMaxPaddingSize,
rtp_sender_->TimeToSendPadding(49, PacedPacketInfo()));
PacketOptions options;
EXPECT_CALL(transport,
SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize, _))
.WillOnce(
testing::DoAll(testing::SaveArg<2>(&options), testing::Return(true)));
EXPECT_EQ(kPayloadSizes[0],
rtp_sender_->TimeToSendPadding(500, PacedPacketInfo()));
EXPECT_TRUE(options.is_retransmit);
EXPECT_CALL(transport, SendRtp(_,
kPayloadSizes[kNumPayloadSizes - 1] +
rtp_header_len + kRtxHeaderSize,
_))
.WillOnce(testing::Return(true));
options.is_retransmit = false;
EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
.WillOnce(
testing::DoAll(testing::SaveArg<2>(&options), testing::Return(true)));
EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize,
rtp_sender_->TimeToSendPadding(999, PacedPacketInfo()));
EXPECT_FALSE(options.is_retransmit);
}
TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
const char payload_name[] = "GENERIC";
const uint8_t payload_type = 127;
PlayoutDelayOracle playout_delay_oracle;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
&playout_delay_oracle, nullptr, false,
FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
uint8_t payload[] = {47, 11, 32, 93, 89};
// Send keyframe
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
auto sent_payload = transport_.last_sent_packet().payload();
uint8_t generic_header = sent_payload[0];
EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(payload));
// Send delta frame
payload[0] = 13;
payload[1] = 42;
payload[4] = 13;
ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
sent_payload = transport_.last_sent_packet().payload();
generic_header = sent_payload[0];
EXPECT_FALSE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(payload));
}
TEST_P(RtpSenderTest, SendFlexfecPackets) {
constexpr uint32_t kTimestamp = 1234;
constexpr int kMediaPayloadType = 127;
constexpr int kFlexfecPayloadType = 118;
constexpr uint32_t kMediaSsrc = 1234;
constexpr uint32_t kFlexfecSsrc = 5678;
const std::vector<RtpExtension> kNoRtpExtensions;
const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes,
nullptr /* rtp_state */, &fake_clock_);
// Reset |rtp_sender_| to use FlexFEC.
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, &mock_paced_sender_, kFlexfecSsrc,
&seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
&send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetStorePacketsStatus(true, 10);
PlayoutDelayOracle playout_delay_oracle;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
&flexfec_sender, &playout_delay_oracle,
nullptr, false, FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
params.fec_rate = 15;
params.max_fec_frames = 1;
params.fec_mask_type = kFecMaskRandom;
rtp_sender_video.SetFecParameters(params, params);
EXPECT_CALL(mock_paced_sender_,
InsertPacket(RtpPacketSender::kLowPriority, kMediaSsrc, kSeqNum,
_, _, false));
uint16_t flexfec_seq_num;
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kLowPriority,
kFlexfecSsrc, _, _, _, false))
.WillOnce(testing::SaveArg<2>(&flexfec_seq_num));
RTPVideoHeader video_header;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kMediaPayloadType, kTimestamp,
fake_clock_.TimeInMilliseconds(), kPayloadData, sizeof(kPayloadData),
nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(2);
EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum,
fake_clock_.TimeInMilliseconds(),
false, PacedPacketInfo()));
EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kFlexfecSsrc, flexfec_seq_num,
fake_clock_.TimeInMilliseconds(),
false, PacedPacketInfo()));
ASSERT_EQ(2, transport_.packets_sent());
const RtpPacketReceived& media_packet = transport_.sent_packets_[0];
EXPECT_EQ(kMediaPayloadType, media_packet.PayloadType());
EXPECT_EQ(kSeqNum, media_packet.SequenceNumber());
EXPECT_EQ(kMediaSsrc, media_packet.Ssrc());
const RtpPacketReceived& flexfec_packet = transport_.sent_packets_[1];
EXPECT_EQ(kFlexfecPayloadType, flexfec_packet.PayloadType());
EXPECT_EQ(flexfec_seq_num, flexfec_packet.SequenceNumber());
EXPECT_EQ(kFlexfecSsrc, flexfec_packet.Ssrc());
}
// TODO(ilnik): because of webrtc:7859. Once FEC moved below pacer, this test
// should be removed.
TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
constexpr uint32_t kTimestamp = 1234;
const int64_t kCaptureTimeMs = fake_clock_.TimeInMilliseconds();
constexpr int kMediaPayloadType = 127;
constexpr int kFlexfecPayloadType = 118;
constexpr uint32_t kMediaSsrc = 1234;
constexpr uint32_t kFlexfecSsrc = 5678;
const std::vector<RtpExtension> kNoRtpExtensions;
const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes,
nullptr /* rtp_state */, &fake_clock_);
// Reset |rtp_sender_| to use FlexFEC.
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, &mock_paced_sender_,
flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetStorePacketsStatus(true, 10);
PlayoutDelayOracle playout_delay_oracle;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
&flexfec_sender, &playout_delay_oracle,
nullptr, false, FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
// Need extension to be registered for timing frames to be sent.
ASSERT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoTiming, kVideoTimingExtensionId));
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
params.fec_rate = 15;
params.max_fec_frames = 1;
params.fec_mask_type = kFecMaskRandom;
rtp_sender_video.SetFecParameters(params, params);
EXPECT_CALL(mock_paced_sender_,
InsertPacket(RtpPacketSender::kLowPriority, kMediaSsrc, kSeqNum,
_, _, false));
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kLowPriority,
kFlexfecSsrc, _, _, _, false))
.Times(0); // Not called because packet should not be protected.
RTPVideoHeader video_header;
video_header.video_timing.flags = VideoSendTiming::kTriggeredByTimer;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kMediaPayloadType, kTimestamp, kCaptureTimeMs,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(1);
EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum,
fake_clock_.TimeInMilliseconds(),
false, PacedPacketInfo()));
ASSERT_EQ(1, transport_.packets_sent());
const RtpPacketReceived& media_packet = transport_.sent_packets_[0];
EXPECT_EQ(kMediaPayloadType, media_packet.PayloadType());
EXPECT_EQ(kSeqNum, media_packet.SequenceNumber());
EXPECT_EQ(kMediaSsrc, media_packet.Ssrc());
// Now try to send not a timing frame.
uint16_t flexfec_seq_num;
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kLowPriority,
kFlexfecSsrc, _, _, _, false))
.WillOnce(testing::SaveArg<2>(&flexfec_seq_num));
EXPECT_CALL(mock_paced_sender_,
InsertPacket(RtpPacketSender::kLowPriority, kMediaSsrc,
kSeqNum + 1, _, _, false));
video_header.video_timing.flags = VideoSendTiming::kInvalid;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kMediaPayloadType, kTimestamp + 1, kCaptureTimeMs + 1,
kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
kDefaultExpectedRetransmissionTimeMs));
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(2);
EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum + 1,
fake_clock_.TimeInMilliseconds(),
false, PacedPacketInfo()));
EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kFlexfecSsrc, flexfec_seq_num,
fake_clock_.TimeInMilliseconds(),
false, PacedPacketInfo()));
ASSERT_EQ(3, transport_.packets_sent());
const RtpPacketReceived& media_packet2 = transport_.sent_packets_[1];
EXPECT_EQ(kMediaPayloadType, media_packet2.PayloadType());
EXPECT_EQ(kSeqNum + 1, media_packet2.SequenceNumber());
EXPECT_EQ(kMediaSsrc, media_packet2.Ssrc());
const RtpPacketReceived& flexfec_packet = transport_.sent_packets_[2];
EXPECT_EQ(kFlexfecPayloadType, flexfec_packet.PayloadType());
EXPECT_EQ(flexfec_seq_num, flexfec_packet.SequenceNumber());
EXPECT_EQ(kFlexfecSsrc, flexfec_packet.Ssrc());
}
TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
constexpr uint32_t kTimestamp = 1234;
constexpr int kMediaPayloadType = 127;
constexpr int kFlexfecPayloadType = 118;
constexpr uint32_t kMediaSsrc = 1234;
constexpr uint32_t kFlexfecSsrc = 5678;
const std::vector<RtpExtension> kNoRtpExtensions;
const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes,
nullptr /* rtp_state */, &fake_clock_);
// Reset |rtp_sender_| to use FlexFEC.
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, nullptr, flexfec_sender.ssrc(),
&seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
&send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
PlayoutDelayOracle playout_delay_oracle;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
&flexfec_sender, &playout_delay_oracle,
nullptr, false, FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
params.fec_rate = 15;
params.max_fec_frames = 1;
params.fec_mask_type = kFecMaskRandom;
rtp_sender_video.SetFecParameters(params, params);
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(2);
RTPVideoHeader video_header;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kMediaPayloadType, kTimestamp,
fake_clock_.TimeInMilliseconds(), kPayloadData, sizeof(kPayloadData),
nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
ASSERT_EQ(2, transport_.packets_sent());
const RtpPacketReceived& media_packet = transport_.sent_packets_[0];
EXPECT_EQ(kMediaPayloadType, media_packet.PayloadType());
EXPECT_EQ(kMediaSsrc, media_packet.Ssrc());
const RtpPacketReceived& flexfec_packet = transport_.sent_packets_[1];
EXPECT_EQ(kFlexfecPayloadType, flexfec_packet.PayloadType());
EXPECT_EQ(kFlexfecSsrc, flexfec_packet.Ssrc());
}
// Test that the MID header extension is included on sent packets when
// configured.
TEST_P(RtpSenderTestWithoutPacer, MidIncludedOnSentPackets) {
const char kMid[] = "mid";
// Register MID header extension and set the MID for the RTPSender.
rtp_sender_->SetSendingMediaStatus(false);
rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionMid, kMidExtensionId);
rtp_sender_->SetMid(kMid);
rtp_sender_->SetSendingMediaStatus(true);
// Send a couple packets.
SendGenericPacket();
SendGenericPacket();
// Expect both packets to have the MID set.
ASSERT_EQ(2u, transport_.sent_packets_.size());
for (const RtpPacketReceived& packet : transport_.sent_packets_) {
std::string mid;
ASSERT_TRUE(packet.GetExtension<RtpMid>(&mid));
EXPECT_EQ(kMid, mid);
}
}
TEST_P(RtpSenderTestWithoutPacer, RidIncludedOnSentPackets) {
const char kRid[] = "f";
rtp_sender_->SetSendingMediaStatus(false);
rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionRtpStreamId,
kRidExtensionId);
rtp_sender_->SetRid(kRid);
rtp_sender_->SetSendingMediaStatus(true);
SendGenericPacket();
ASSERT_EQ(1u, transport_.sent_packets_.size());
const RtpPacketReceived& packet = transport_.sent_packets_[0];
std::string rid;
ASSERT_TRUE(packet.GetExtension<RtpStreamId>(&rid));
EXPECT_EQ(kRid, rid);
}
TEST_P(RtpSenderTestWithoutPacer, RidIncludedOnRtxSentPackets) {
const char kRid[] = "f";
rtp_sender_->SetSendingMediaStatus(false);
rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionRtpStreamId,
kRidExtensionId);
rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionRepairedRtpStreamId,
kRepairedRidExtensionId);
rtp_sender_->SetRid(kRid);
rtp_sender_->SetSendingMediaStatus(true);
rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
rtp_sender_->SetRtxSsrc(1234);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender_->SetStorePacketsStatus(true, 10);
SendGenericPacket();
ASSERT_EQ(1u, transport_.sent_packets_.size());
const RtpPacketReceived& packet = transport_.sent_packets_[0];
std::string rid;
ASSERT_TRUE(packet.GetExtension<RtpStreamId>(&rid));
EXPECT_EQ(kRid, rid);
rid = kNoRid;
EXPECT_FALSE(packet.GetExtension<RepairedRtpStreamId>(&rid));
uint16_t packet_id = packet.SequenceNumber();
rtp_sender_->ReSendPacket(packet_id);
ASSERT_EQ(2u, transport_.sent_packets_.size());
const RtpPacketReceived& rtx_packet = transport_.sent_packets_[1];
ASSERT_TRUE(rtx_packet.GetExtension<RepairedRtpStreamId>(&rid));
EXPECT_EQ(kRid, rid);
EXPECT_FALSE(rtx_packet.HasExtension<RtpStreamId>());
}
TEST_P(RtpSenderTest, FecOverheadRate) {
constexpr uint32_t kTimestamp = 1234;
constexpr int kMediaPayloadType = 127;
constexpr int kFlexfecPayloadType = 118;
constexpr uint32_t kMediaSsrc = 1234;
constexpr uint32_t kFlexfecSsrc = 5678;
const std::vector<RtpExtension> kNoRtpExtensions;
const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes,
nullptr /* rtp_state */, &fake_clock_);
// Reset |rtp_sender_| to use FlexFEC.
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, &mock_paced_sender_,
flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
PlayoutDelayOracle playout_delay_oracle;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
&flexfec_sender, &playout_delay_oracle,
nullptr, false, FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
params.fec_rate = 15;
params.max_fec_frames = 1;
params.fec_mask_type = kFecMaskRandom;
rtp_sender_video.SetFecParameters(params, params);
constexpr size_t kNumMediaPackets = 10;
constexpr size_t kNumFecPackets = kNumMediaPackets;
constexpr int64_t kTimeBetweenPacketsMs = 10;
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, false))
.Times(kNumMediaPackets + kNumFecPackets);
for (size_t i = 0; i < kNumMediaPackets; ++i) {
RTPVideoHeader video_header;
EXPECT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, kMediaPayloadType, kTimestamp,
fake_clock_.TimeInMilliseconds(), kPayloadData, sizeof(kPayloadData),
nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
fake_clock_.AdvanceTimeMilliseconds(kTimeBetweenPacketsMs);
}
constexpr size_t kRtpHeaderLength = 12;
constexpr size_t kFlexfecHeaderLength = 20;
constexpr size_t kGenericCodecHeaderLength = 1;
constexpr size_t kPayloadLength = sizeof(kPayloadData);
constexpr size_t kPacketLength = kRtpHeaderLength + kFlexfecHeaderLength +
kGenericCodecHeaderLength + kPayloadLength;
EXPECT_NEAR(kNumFecPackets * kPacketLength * 8 /
(kNumFecPackets * kTimeBetweenPacketsMs / 1000.0f),
rtp_sender_video.FecOverheadRate(), 500);
}
TEST_P(RtpSenderTest, BitrateCallbacks) {
class TestCallback : public BitrateStatisticsObserver {
public:
TestCallback()
: BitrateStatisticsObserver(),
num_calls_(0),
ssrc_(0),
total_bitrate_(0),
retransmit_bitrate_(0) {}
~TestCallback() override = default;
void Notify(uint32_t total_bitrate,
uint32_t retransmit_bitrate,
uint32_t ssrc) override {
++num_calls_;
ssrc_ = ssrc;
total_bitrate_ = total_bitrate;
retransmit_bitrate_ = retransmit_bitrate;
}
uint32_t num_calls_;
uint32_t ssrc_;
uint32_t total_bitrate_;
uint32_t retransmit_bitrate_;
} callback;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
nullptr, nullptr, &callback, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, nullptr, false, nullptr,
false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
PlayoutDelayOracle playout_delay_oracle;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
&playout_delay_oracle, nullptr, false,
FieldTrialBasedConfig());
const char payload_name[] = "GENERIC";
const uint8_t payload_type = 127;
rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
// Simulate kNumPackets sent with kPacketInterval ms intervals, with the
// number of packets selected so that we fill (but don't overflow) the one
// second averaging window.
const uint32_t kWindowSizeMs = 1000;
const uint32_t kPacketInterval = 20;
const uint32_t kNumPackets =
(kWindowSizeMs - kPacketInterval) / kPacketInterval;
// Overhead = 12 bytes RTP header + 1 byte generic header.
const uint32_t kPacketOverhead = 13;
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
// Initial process call so we get a new time window.
rtp_sender_->ProcessBitrate();
// Send a few frames.
RTPVideoHeader video_header;
for (uint32_t i = 0; i < kNumPackets; ++i) {
ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
}
rtp_sender_->ProcessBitrate();
// We get one call for every stats updated, thus two calls since both the
// stream stats and the retransmit stats are updated once.
EXPECT_EQ(2u, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
const uint32_t kTotalPacketSize = kPacketOverhead + sizeof(payload);
// Bitrate measured over delta between last and first timestamp, plus one.
const uint32_t kExpectedWindowMs = kNumPackets * kPacketInterval + 1;
const uint32_t kExpectedBitsAccumulated = kTotalPacketSize * kNumPackets * 8;
const uint32_t kExpectedRateBps =
(kExpectedBitsAccumulated * 1000 + (kExpectedWindowMs / 2)) /
kExpectedWindowMs;
EXPECT_EQ(kExpectedRateBps, callback.total_bitrate_);
rtp_sender_.reset();
}
TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
class TestCallback : public StreamDataCountersCallback {
public:
TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
~TestCallback() override = default;
void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) override {
ssrc_ = ssrc;
counters_ = counters;
}
uint32_t ssrc_;
StreamDataCounters counters_;
void MatchPacketCounter(const RtpPacketCounter& expected,
const RtpPacketCounter& actual) {
EXPECT_EQ(expected.payload_bytes, actual.payload_bytes);
EXPECT_EQ(expected.header_bytes, actual.header_bytes);
EXPECT_EQ(expected.padding_bytes, actual.padding_bytes);
EXPECT_EQ(expected.packets, actual.packets);
}
void Matches(uint32_t ssrc, const StreamDataCounters& counters) {
EXPECT_EQ(ssrc, ssrc_);
MatchPacketCounter(counters.transmitted, counters_.transmitted);
MatchPacketCounter(counters.retransmitted, counters_.retransmitted);
EXPECT_EQ(counters.fec.packets, counters_.fec.packets);
}
} callback;
const uint8_t kRedPayloadType = 96;
const uint8_t kUlpfecPayloadType = 97;
const char payload_name[] = "GENERIC";
const uint8_t payload_type = 127;
PlayoutDelayOracle playout_delay_oracle;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
&playout_delay_oracle, nullptr, false,
FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
rtp_sender_->RegisterRtpStatisticsCallback(&callback);
// Send a frame.
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
StreamDataCounters expected;
expected.transmitted.payload_bytes = 6;
expected.transmitted.header_bytes = 12;
expected.transmitted.padding_bytes = 0;
expected.transmitted.packets = 1;
expected.retransmitted.payload_bytes = 0;
expected.retransmitted.header_bytes = 0;
expected.retransmitted.padding_bytes = 0;
expected.retransmitted.packets = 0;
expected.fec.packets = 0;
callback.Matches(ssrc, expected);
// Retransmit a frame.
uint16_t seqno = rtp_sender_->SequenceNumber() - 1;
rtp_sender_->ReSendPacket(seqno);
expected.transmitted.payload_bytes = 12;
expected.transmitted.header_bytes = 24;
expected.transmitted.packets = 2;
expected.retransmitted.payload_bytes = 6;
expected.retransmitted.header_bytes = 12;
expected.retransmitted.padding_bytes = 0;
expected.retransmitted.packets = 1;
callback.Matches(ssrc, expected);
// Send padding.
rtp_sender_->TimeToSendPadding(kMaxPaddingSize, PacedPacketInfo());
expected.transmitted.payload_bytes = 12;
expected.transmitted.header_bytes = 36;
expected.transmitted.padding_bytes = kMaxPaddingSize;
expected.transmitted.packets = 3;
callback.Matches(ssrc, expected);
// Send ULPFEC.
rtp_sender_video.SetUlpfecConfig(kRedPayloadType, kUlpfecPayloadType);
FecProtectionParams fec_params;
fec_params.fec_mask_type = kFecMaskRandom;
fec_params.fec_rate = 1;
fec_params.max_fec_frames = 1;
rtp_sender_video.SetFecParameters(fec_params, fec_params);
ASSERT_TRUE(rtp_sender_video.SendVideo(
kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
expected.transmitted.payload_bytes = 40;
expected.transmitted.header_bytes = 60;
expected.transmitted.packets = 5;
expected.fec.packets = 1;
callback.Matches(ssrc, expected);
rtp_sender_->RegisterRtpStatisticsCallback(nullptr);
}
TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
// XXX const char* kPayloadName = "GENERIC";
const uint8_t kPayloadType = 127;
rtp_sender_->SetSSRC(1234);
rtp_sender_->SetRtxSsrc(4321);
rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
SendGenericPacket();
// Will send 2 full-size padding packets.
rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());
rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
// Payload
EXPECT_GT(rtp_stats.first_packet_time_ms, -1);
EXPECT_EQ(rtp_stats.transmitted.payload_bytes, sizeof(kPayloadData));
EXPECT_EQ(rtp_stats.transmitted.header_bytes, 12u);
EXPECT_EQ(rtp_stats.transmitted.padding_bytes, 0u);
EXPECT_EQ(rtx_stats.transmitted.payload_bytes, 0u);
EXPECT_EQ(rtx_stats.transmitted.header_bytes, 24u);
EXPECT_EQ(rtx_stats.transmitted.padding_bytes, 2 * kMaxPaddingSize);
EXPECT_EQ(rtp_stats.transmitted.TotalBytes(),
rtp_stats.transmitted.payload_bytes +
rtp_stats.transmitted.header_bytes +
rtp_stats.transmitted.padding_bytes);
EXPECT_EQ(rtx_stats.transmitted.TotalBytes(),
rtx_stats.transmitted.payload_bytes +
rtx_stats.transmitted.header_bytes +
rtx_stats.transmitted.padding_bytes);
EXPECT_EQ(
transport_.total_bytes_sent_,
rtp_stats.transmitted.TotalBytes() + rtx_stats.transmitted.TotalBytes());
}
TEST_P(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
const int32_t kPacketSize = 1400;
const int32_t kNumPackets = 30;
retransmission_rate_limiter_.SetMaxRate(kPacketSize * kNumPackets * 8);
rtp_sender_->SetStorePacketsStatus(true, kNumPackets);
const uint16_t kStartSequenceNumber = rtp_sender_->SequenceNumber();
std::vector<uint16_t> sequence_numbers;
for (int32_t i = 0; i < kNumPackets; ++i) {
sequence_numbers.push_back(kStartSequenceNumber + i);
fake_clock_.AdvanceTimeMilliseconds(1);
SendPacket(fake_clock_.TimeInMilliseconds(), kPacketSize);
}
EXPECT_EQ(kNumPackets, transport_.packets_sent());
fake_clock_.AdvanceTimeMilliseconds(1000 - kNumPackets);
// Resending should work - brings the bandwidth up to the limit.
// NACK bitrate is capped to the same bitrate as the encoder, since the max
// protection overhead is 50% (see MediaOptimization::SetTargetRates).
rtp_sender_->OnReceivedNack(sequence_numbers, 0);
EXPECT_EQ(kNumPackets * 2, transport_.packets_sent());
// Must be at least 5ms in between retransmission attempts.
fake_clock_.AdvanceTimeMilliseconds(5);
// Resending should not work, bandwidth exceeded.
rtp_sender_->OnReceivedNack(sequence_numbers, 0);
EXPECT_EQ(kNumPackets * 2, transport_.packets_sent());
}
TEST_P(RtpSenderTest, OnOverheadChanged) {
MockOverheadObserver mock_overhead_observer;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer,
false, nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
// RTP overhead is 12B.
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(12)).Times(1);
SendGenericPacket();
rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
// TransmissionTimeOffset extension has a size of 8B.
// 12B + 8B = 20B
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(20)).Times(1);
SendGenericPacket();
}
TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
MockOverheadObserver mock_overhead_observer;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer,
false, nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
SendGenericPacket();
SendGenericPacket();
}
TEST_P(RtpSenderTest, SendsKeepAlive) {
MockTransport transport;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport, nullptr, absl::nullopt,
nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, nullptr, false,
nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
const uint8_t kKeepalivePayloadType = 20;
RTC_CHECK_NE(kKeepalivePayloadType, kPayload);
EXPECT_CALL(transport, SendRtp(_, _, _))
.WillOnce(
Invoke([&kKeepalivePayloadType](const uint8_t* packet, size_t len,
const PacketOptions& options) {
webrtc::RTPHeader rtp_header;
RtpUtility::RtpHeaderParser parser(packet, len);
EXPECT_TRUE(parser.Parse(&rtp_header, nullptr));
EXPECT_FALSE(rtp_header.markerBit);
EXPECT_EQ(0U, rtp_header.paddingLength);
EXPECT_EQ(kKeepalivePayloadType, rtp_header.payloadType);
EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber);
EXPECT_EQ(kSsrc, rtp_header.ssrc);
EXPECT_EQ(0u, len - rtp_header.headerLength);
return true;
}));
rtp_sender_->SendKeepAlive(kKeepalivePayloadType);
EXPECT_EQ(kSeqNum + 1, rtp_sender_->SequenceNumber());
}
INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
RtpSenderTest,
::testing::Bool());
INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
RtpSenderTestWithoutPacer,
::testing::Bool());
} // namespace webrtc
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