0b9e29c87d
Also move files out of media_file/source. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=asapersson@webrtc.org, perkj@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1435093002 . Cr-Commit-Position: refs/heads/master@{#10647}
173 lines
7.2 KiB
C++
173 lines
7.2 KiB
C++
/*
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* Copyright (c) 2015 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 <list>
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#include "webrtc/base/checks.h"
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#include "testing/gmock/include/gmock/gmock.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/pacing/packet_router.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
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#include "webrtc/base/scoped_ptr.h"
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using ::testing::_;
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using ::testing::AnyNumber;
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using ::testing::NiceMock;
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using ::testing::Return;
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namespace webrtc {
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class PacketRouterTest : public ::testing::Test {
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public:
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PacketRouterTest() : packet_router_(new PacketRouter()) {}
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protected:
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const rtc::scoped_ptr<PacketRouter> packet_router_;
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};
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TEST_F(PacketRouterTest, TimeToSendPacket) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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packet_router_->AddRtpModule(&rtp_1);
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packet_router_->AddRtpModule(&rtp_2);
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const uint16_t kSsrc1 = 1234;
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uint16_t sequence_number = 17;
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uint64_t timestamp = 7890;
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bool retransmission = false;
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// Send on the first module by letting rtp_1 be sending with correct ssrc.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
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EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc1, sequence_number, timestamp,
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retransmission))
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.Times(1)
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.WillOnce(Return(true));
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
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timestamp, retransmission));
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// Send on the second module by letting rtp_2 be sending, but not rtp_1.
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++sequence_number;
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timestamp += 30;
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retransmission = true;
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const uint16_t kSsrc2 = 4567;
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
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EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_CALL(rtp_2, TimeToSendPacket(kSsrc2, sequence_number, timestamp,
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retransmission))
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.Times(1)
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.WillOnce(Return(true));
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc2, sequence_number,
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timestamp, retransmission));
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// No module is sending, hence no packet should be sent.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
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timestamp, retransmission));
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// Add a packet with incorrect ssrc and test it's dropped in the router.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
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EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1 + kSsrc2, sequence_number,
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timestamp, retransmission));
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packet_router_->RemoveRtpModule(&rtp_1);
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// rtp_1 has been removed, try sending a packet on that ssrc and make sure
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// it is dropped as expected by not expecting any calls to rtp_1.
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
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timestamp, retransmission));
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packet_router_->RemoveRtpModule(&rtp_2);
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}
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TEST_F(PacketRouterTest, TimeToSendPadding) {
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const uint16_t kSsrc1 = 1234;
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const uint16_t kSsrc2 = 4567;
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MockRtpRtcp rtp_1;
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EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
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MockRtpRtcp rtp_2;
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EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
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packet_router_->AddRtpModule(&rtp_1);
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packet_router_->AddRtpModule(&rtp_2);
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// Default configuration, sending padding on all modules sending media,
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// ordered by SSRC.
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const size_t requested_padding_bytes = 1000;
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const size_t sent_padding_bytes = 890;
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
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.Times(1)
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.WillOnce(Return(sent_padding_bytes));
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2,
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TimeToSendPadding(requested_padding_bytes - sent_padding_bytes))
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.Times(1)
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.WillOnce(Return(requested_padding_bytes - sent_padding_bytes));
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EXPECT_EQ(requested_padding_bytes,
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packet_router_->TimeToSendPadding(requested_padding_bytes));
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// Let only the second module be sending and verify the padding request is
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// routed there.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0);
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, TimeToSendPadding(_))
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.Times(1)
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.WillOnce(Return(sent_padding_bytes));
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EXPECT_EQ(sent_padding_bytes,
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packet_router_->TimeToSendPadding(requested_padding_bytes));
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// No sending module at all.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0);
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0);
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EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
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packet_router_->RemoveRtpModule(&rtp_1);
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// rtp_1 has been removed, try sending padding and make sure rtp_1 isn't asked
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// to send by not expecting any calls. Instead verify rtp_2 is called.
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes)).Times(1);
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EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
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packet_router_->RemoveRtpModule(&rtp_2);
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}
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TEST_F(PacketRouterTest, AllocateSequenceNumbers) {
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const uint16_t kStartSeq = 0xFFF0;
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const size_t kNumPackets = 32;
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packet_router_->SetTransportWideSequenceNumber(kStartSeq - 1);
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for (size_t i = 0; i < kNumPackets; ++i) {
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uint16_t seq = packet_router_->AllocateSequenceNumber();
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uint32_t expected_unwrapped_seq = static_cast<uint32_t>(kStartSeq) + i;
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EXPECT_EQ(static_cast<uint16_t>(expected_unwrapped_seq & 0xFFFF), seq);
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}
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}
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} // namespace webrtc
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