| /* |
| * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "modules/pacing/task_queue_paced_sender.h" |
| |
| #include <list> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "modules/pacing/packet_router.h" |
| #include "modules/utility/include/mock/mock_process_thread.h" |
| #include "test/field_trial.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| #include "test/time_controller/simulated_time_controller.h" |
| |
| using ::testing::_; |
| using ::testing::Return; |
| using ::testing::SaveArg; |
| |
| namespace webrtc { |
| namespace { |
| constexpr uint32_t kAudioSsrc = 12345; |
| constexpr uint32_t kVideoSsrc = 234565; |
| constexpr uint32_t kVideoRtxSsrc = 34567; |
| constexpr uint32_t kFlexFecSsrc = 45678; |
| constexpr size_t kDefaultPacketSize = 1234; |
| |
| class MockPacketRouter : public PacketRouter { |
| public: |
| MOCK_METHOD2(SendPacket, |
| void(std::unique_ptr<RtpPacketToSend> packet, |
| const PacedPacketInfo& cluster_info)); |
| MOCK_METHOD1( |
| GeneratePadding, |
| std::vector<std::unique_ptr<RtpPacketToSend>>(size_t target_size_bytes)); |
| }; |
| } // namespace |
| |
| namespace test { |
| |
| class TaskQueuePacedSenderTest : public ::testing::Test { |
| public: |
| TaskQueuePacedSenderTest() |
| : time_controller_(Timestamp::Millis(1234)), |
| pacer_(time_controller_.GetClock(), |
| &packet_router_, |
| /*event_log=*/nullptr, |
| /*field_trials=*/nullptr, |
| time_controller_.GetTaskQueueFactory()) {} |
| |
| protected: |
| std::unique_ptr<RtpPacketToSend> BuildRtpPacket(RtpPacketMediaType type) { |
| auto packet = std::make_unique<RtpPacketToSend>(nullptr); |
| packet->set_packet_type(type); |
| switch (type) { |
| case RtpPacketMediaType::kAudio: |
| packet->SetSsrc(kAudioSsrc); |
| break; |
| case RtpPacketMediaType::kVideo: |
| packet->SetSsrc(kVideoSsrc); |
| break; |
| case RtpPacketMediaType::kRetransmission: |
| case RtpPacketMediaType::kPadding: |
| packet->SetSsrc(kVideoRtxSsrc); |
| break; |
| case RtpPacketMediaType::kForwardErrorCorrection: |
| packet->SetSsrc(kFlexFecSsrc); |
| break; |
| } |
| |
| packet->SetPayloadSize(kDefaultPacketSize); |
| return packet; |
| } |
| |
| std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePackets( |
| RtpPacketMediaType type, |
| size_t num_packets) { |
| std::vector<std::unique_ptr<RtpPacketToSend>> packets; |
| for (size_t i = 0; i < num_packets; ++i) { |
| packets.push_back(BuildRtpPacket(type)); |
| } |
| return packets; |
| } |
| |
| Timestamp CurrentTime() { return time_controller_.GetClock()->CurrentTime(); } |
| |
| GlobalSimulatedTimeController time_controller_; |
| MockPacketRouter packet_router_; |
| TaskQueuePacedSender pacer_; |
| }; |
| |
| TEST_F(TaskQueuePacedSenderTest, PacesPackets) { |
| // Insert a number of packets, covering one second. |
| static constexpr size_t kPacketsToSend = 42; |
| pacer_.SetPacingRates( |
| DataRate::BitsPerSec(kDefaultPacketSize * 8 * kPacketsToSend), |
| DataRate::Zero()); |
| pacer_.EnqueuePackets( |
| GeneratePackets(RtpPacketMediaType::kVideo, kPacketsToSend)); |
| |
| // Expect all of them to be sent. |
| size_t packets_sent = 0; |
| Timestamp end_time = Timestamp::PlusInfinity(); |
| EXPECT_CALL(packet_router_, SendPacket) |
| .WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet, |
| const PacedPacketInfo& cluster_info) { |
| ++packets_sent; |
| if (packets_sent == kPacketsToSend) { |
| end_time = time_controller_.GetClock()->CurrentTime(); |
| } |
| }); |
| |
| const Timestamp start_time = time_controller_.GetClock()->CurrentTime(); |
| |
| // Packets should be sent over a period of close to 1s. Expect a little lower |
| // than this since initial probing is a bit quicker. |
| time_controller_.AdvanceTime(TimeDelta::Seconds(1)); |
| EXPECT_EQ(packets_sent, kPacketsToSend); |
| ASSERT_TRUE(end_time.IsFinite()); |
| EXPECT_NEAR((end_time - start_time).ms<double>(), 1000.0, 50.0); |
| } |
| |
| TEST_F(TaskQueuePacedSenderTest, ReschedulesProcessOnRateChange) { |
| // Insert a number of packets to be sent 200ms apart. |
| const size_t kPacketsPerSecond = 5; |
| const DataRate kPacingRate = |
| DataRate::BitsPerSec(kDefaultPacketSize * 8 * kPacketsPerSecond); |
| pacer_.SetPacingRates(kPacingRate, DataRate::Zero()); |
| |
| // Send some initial packets to be rid of any probes. |
| EXPECT_CALL(packet_router_, SendPacket).Times(kPacketsPerSecond); |
| pacer_.EnqueuePackets( |
| GeneratePackets(RtpPacketMediaType::kVideo, kPacketsPerSecond)); |
| time_controller_.AdvanceTime(TimeDelta::Seconds(1)); |
| |
| // Insert three packets, and record send time of each of them. |
| // After the second packet is sent, double the send rate so we can |
| // check the third packets is sent after half the wait time. |
| Timestamp first_packet_time = Timestamp::MinusInfinity(); |
| Timestamp second_packet_time = Timestamp::MinusInfinity(); |
| Timestamp third_packet_time = Timestamp::MinusInfinity(); |
| |
| EXPECT_CALL(packet_router_, SendPacket) |
| .Times(3) |
| .WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet, |
| const PacedPacketInfo& cluster_info) { |
| if (first_packet_time.IsInfinite()) { |
| first_packet_time = CurrentTime(); |
| } else if (second_packet_time.IsInfinite()) { |
| second_packet_time = CurrentTime(); |
| pacer_.SetPacingRates(2 * kPacingRate, DataRate::Zero()); |
| } else { |
| third_packet_time = CurrentTime(); |
| } |
| }); |
| |
| pacer_.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 3)); |
| time_controller_.AdvanceTime(TimeDelta::Millis(500)); |
| ASSERT_TRUE(third_packet_time.IsFinite()); |
| EXPECT_NEAR((second_packet_time - first_packet_time).ms<double>(), 200.0, |
| 1.0); |
| EXPECT_NEAR((third_packet_time - second_packet_time).ms<double>(), 100.0, |
| 1.0); |
| } |
| |
| TEST_F(TaskQueuePacedSenderTest, SendsAudioImmediately) { |
| const DataRate kPacingDataRate = DataRate::KilobitsPerSec(125); |
| const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize); |
| const TimeDelta kPacketPacingTime = kPacketSize / kPacingDataRate; |
| |
| pacer_.SetPacingRates(kPacingDataRate, DataRate::Zero()); |
| |
| // Add some initial video packets, only one should be sent. |
| EXPECT_CALL(packet_router_, SendPacket); |
| pacer_.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 10)); |
| time_controller_.AdvanceTime(TimeDelta::Zero()); |
| ::testing::Mock::VerifyAndClearExpectations(&packet_router_); |
| |
| // Advance time, but still before next packet should be sent. |
| time_controller_.AdvanceTime(kPacketPacingTime / 2); |
| |
| // Insert an audio packet, it should be sent immediately. |
| EXPECT_CALL(packet_router_, SendPacket); |
| pacer_.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kAudio, 1)); |
| time_controller_.AdvanceTime(TimeDelta::Zero()); |
| ::testing::Mock::VerifyAndClearExpectations(&packet_router_); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |