317a1f09ed
WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
118 lines
3.7 KiB
C++
118 lines
3.7 KiB
C++
/*
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* Copyright (c) 2012 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/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 "system_wrappers/include/clock.h"
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#include "system_wrappers/include/field_trial.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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using ::testing::_;
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using ::testing::Return;
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using ::testing::SaveArg;
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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 = 234;
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} // namespace
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namespace webrtc {
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namespace test {
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// Mock callback implementing the raw api.
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class MockCallback : 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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std::unique_ptr<RtpPacketToSend> BuildRtpPacket(RtpPacketToSend::Type 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 RtpPacketToSend::Type::kAudio:
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packet->SetSsrc(kAudioSsrc);
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break;
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case RtpPacketToSend::Type::kVideo:
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packet->SetSsrc(kVideoSsrc);
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break;
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case RtpPacketToSend::Type::kRetransmission:
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case RtpPacketToSend::Type::kPadding:
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packet->SetSsrc(kVideoRtxSsrc);
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break;
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case RtpPacketToSend::Type::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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TEST(PacedSenderTest, PacesPackets) {
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SimulatedClock clock(0);
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MockCallback callback;
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MockProcessThread process_thread;
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Module* paced_module = nullptr;
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EXPECT_CALL(process_thread, RegisterModule(_, _))
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.WillOnce(SaveArg<0>(&paced_module));
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PacedSender pacer(&clock, &callback, nullptr, nullptr, &process_thread);
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EXPECT_CALL(process_thread, DeRegisterModule(paced_module)).Times(1);
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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(DataRate::bps(kDefaultPacketSize * 8 * kPacketsToSend),
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DataRate::Zero());
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for (size_t i = 0; i < kPacketsToSend; ++i) {
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pacer.EnqueuePacket(BuildRtpPacket(RtpPacketToSend::Type::kVideo));
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}
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// Expect all of them to be sent.
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size_t packets_sent = 0;
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clock.AdvanceTimeMilliseconds(paced_module->TimeUntilNextProcess());
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EXPECT_CALL(callback, SendPacket)
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.WillRepeatedly(
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[&](std::unique_ptr<RtpPacketToSend> packet,
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const PacedPacketInfo& cluster_info) { ++packets_sent; });
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const Timestamp start_time = clock.CurrentTime();
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while (packets_sent < kPacketsToSend) {
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clock.AdvanceTimeMilliseconds(paced_module->TimeUntilNextProcess());
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paced_module->Process();
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}
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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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TimeDelta duration = clock.CurrentTime() - start_time;
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EXPECT_GT(duration, TimeDelta::ms(900));
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}
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} // namespace test
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} // namespace webrtc
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