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This is search and replace change: find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g" git cl format Bug: webrtc:9709 Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30545}
177 lines
6.1 KiB
C++
177 lines
6.1 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/pacing/task_queue_paced_sender.h"
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#include <list>
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#include <memory>
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#include <string>
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#include <utility>
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#include <vector>
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#include "modules/pacing/packet_router.h"
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#include "modules/utility/include/mock/mock_process_thread.h"
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#include "test/field_trial.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include "test/time_controller/simulated_time_controller.h"
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using ::testing::_;
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using ::testing::Return;
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using ::testing::SaveArg;
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namespace webrtc {
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namespace {
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constexpr uint32_t kAudioSsrc = 12345;
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constexpr uint32_t kVideoSsrc = 234565;
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constexpr uint32_t kVideoRtxSsrc = 34567;
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constexpr uint32_t kFlexFecSsrc = 45678;
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constexpr size_t kDefaultPacketSize = 1234;
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class MockPacketRouter : public PacketRouter {
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public:
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MOCK_METHOD2(SendPacket,
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void(std::unique_ptr<RtpPacketToSend> packet,
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const PacedPacketInfo& cluster_info));
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MOCK_METHOD1(
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GeneratePadding,
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std::vector<std::unique_ptr<RtpPacketToSend>>(size_t target_size_bytes));
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};
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} // namespace
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namespace test {
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class TaskQueuePacedSenderTest : public ::testing::Test {
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public:
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TaskQueuePacedSenderTest()
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: time_controller_(Timestamp::Millis(1234)),
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pacer_(time_controller_.GetClock(),
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&packet_router_,
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/*event_log=*/nullptr,
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/*field_trials=*/nullptr,
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time_controller_.GetTaskQueueFactory()) {}
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protected:
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std::unique_ptr<RtpPacketToSend> BuildRtpPacket(RtpPacketMediaType type) {
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auto packet = std::make_unique<RtpPacketToSend>(nullptr);
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packet->set_packet_type(type);
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switch (type) {
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case RtpPacketMediaType::kAudio:
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packet->SetSsrc(kAudioSsrc);
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break;
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case RtpPacketMediaType::kVideo:
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packet->SetSsrc(kVideoSsrc);
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break;
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case RtpPacketMediaType::kRetransmission:
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case RtpPacketMediaType::kPadding:
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packet->SetSsrc(kVideoRtxSsrc);
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break;
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case RtpPacketMediaType::kForwardErrorCorrection:
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packet->SetSsrc(kFlexFecSsrc);
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break;
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}
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packet->SetPayloadSize(kDefaultPacketSize);
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return packet;
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}
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std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePackets(
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RtpPacketMediaType type,
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size_t num_packets) {
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std::vector<std::unique_ptr<RtpPacketToSend>> packets;
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for (size_t i = 0; i < num_packets; ++i) {
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packets.push_back(BuildRtpPacket(type));
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}
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return packets;
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}
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Timestamp CurrentTime() { return time_controller_.GetClock()->CurrentTime(); }
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GlobalSimulatedTimeController time_controller_;
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MockPacketRouter packet_router_;
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TaskQueuePacedSender pacer_;
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};
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TEST_F(TaskQueuePacedSenderTest, PacesPackets) {
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// Insert a number of packets, covering one second.
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static constexpr size_t kPacketsToSend = 42;
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pacer_.SetPacingRates(
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DataRate::BitsPerSec(kDefaultPacketSize * 8 * kPacketsToSend),
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DataRate::Zero());
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pacer_.EnqueuePackets(
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GeneratePackets(RtpPacketMediaType::kVideo, kPacketsToSend));
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// Expect all of them to be sent.
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size_t packets_sent = 0;
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Timestamp end_time = Timestamp::PlusInfinity();
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EXPECT_CALL(packet_router_, SendPacket)
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.WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet,
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const PacedPacketInfo& cluster_info) {
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++packets_sent;
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if (packets_sent == kPacketsToSend) {
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end_time = time_controller_.GetClock()->CurrentTime();
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}
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});
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const Timestamp start_time = time_controller_.GetClock()->CurrentTime();
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// Packets should be sent over a period of close to 1s. Expect a little lower
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// than this since initial probing is a bit quicker.
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time_controller_.AdvanceTime(TimeDelta::Seconds(1));
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EXPECT_EQ(packets_sent, kPacketsToSend);
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ASSERT_TRUE(end_time.IsFinite());
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EXPECT_NEAR((end_time - start_time).ms<double>(), 1000.0, 50.0);
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}
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TEST_F(TaskQueuePacedSenderTest, ReschedulesProcessOnRateChange) {
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// Insert a number of packets to be sent 200ms apart.
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const size_t kPacketsPerSecond = 5;
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const DataRate kPacingRate =
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DataRate::BitsPerSec(kDefaultPacketSize * 8 * kPacketsPerSecond);
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pacer_.SetPacingRates(kPacingRate, DataRate::Zero());
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// Send some initial packets to be rid of any probes.
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EXPECT_CALL(packet_router_, SendPacket).Times(kPacketsPerSecond);
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pacer_.EnqueuePackets(
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GeneratePackets(RtpPacketMediaType::kVideo, kPacketsPerSecond));
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time_controller_.AdvanceTime(TimeDelta::Seconds(1));
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// Insert three packets, and record send time of each of them.
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// After the second packet is sent, double the send rate so we can
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// check the third packets is sent after half the wait time.
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Timestamp first_packet_time = Timestamp::MinusInfinity();
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Timestamp second_packet_time = Timestamp::MinusInfinity();
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Timestamp third_packet_time = Timestamp::MinusInfinity();
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EXPECT_CALL(packet_router_, SendPacket)
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.Times(3)
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.WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet,
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const PacedPacketInfo& cluster_info) {
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if (first_packet_time.IsInfinite()) {
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first_packet_time = CurrentTime();
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} else if (second_packet_time.IsInfinite()) {
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second_packet_time = CurrentTime();
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pacer_.SetPacingRates(2 * kPacingRate, DataRate::Zero());
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} else {
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third_packet_time = CurrentTime();
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}
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});
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pacer_.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 3));
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time_controller_.AdvanceTime(TimeDelta::Millis(500));
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ASSERT_TRUE(third_packet_time.IsFinite());
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EXPECT_NEAR((second_packet_time - first_packet_time).ms<double>(), 200.0,
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1.0);
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EXPECT_NEAR((third_packet_time - second_packet_time).ms<double>(), 100.0,
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1.0);
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}
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} // namespace test
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} // namespace webrtc
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