/* * Copyright (c) 2015 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 #include "webrtc/base/checks.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/pacing/include/packet_router.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/base/scoped_ptr.h" using ::testing::_; using ::testing::AnyNumber; using ::testing::NiceMock; using ::testing::Return; namespace webrtc { class PacketRouterTest : public ::testing::Test { public: PacketRouterTest() : packet_router_(new PacketRouter()) {} protected: const rtc::scoped_ptr packet_router_; }; TEST_F(PacketRouterTest, TimeToSendPacket) { MockRtpRtcp rtp_1; MockRtpRtcp rtp_2; packet_router_->AddRtpModule(&rtp_1); packet_router_->AddRtpModule(&rtp_2); const uint16_t kSsrc1 = 1234; uint16_t sequence_number = 17; uint64_t timestamp = 7890; bool retransmission = false; // Send on the first module by letting rtp_1 be sending with correct ssrc. EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1)); EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc1, sequence_number, timestamp, retransmission)) .Times(1) .WillOnce(Return(true)); EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0); EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number, timestamp, retransmission)); // Send on the second module by letting rtp_2 be sending, but not rtp_1. ++sequence_number; timestamp += 30; retransmission = true; const uint16_t kSsrc2 = 4567; EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false)); EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2)); EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0); EXPECT_CALL(rtp_2, TimeToSendPacket(kSsrc2, sequence_number, timestamp, retransmission)) .Times(1) .WillOnce(Return(true)); EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc2, sequence_number, timestamp, retransmission)); // No module is sending, hence no packet should be sent. EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false)); EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0); EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false)); EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0); EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number, timestamp, retransmission)); // Add a packet with incorrect ssrc and test it's dropped in the router. EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1)); EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2)); EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0); EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0); EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1 + kSsrc2, sequence_number, timestamp, retransmission)); packet_router_->RemoveRtpModule(&rtp_1); // rtp_1 has been removed, try sending a packet on that ssrc and make sure // it is dropped as expected by not expecting any calls to rtp_1. EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2)); EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0); EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number, timestamp, retransmission)); packet_router_->RemoveRtpModule(&rtp_2); } TEST_F(PacketRouterTest, TimeToSendPadding) { MockRtpRtcp rtp_1; MockRtpRtcp rtp_2; packet_router_->AddRtpModule(&rtp_1); packet_router_->AddRtpModule(&rtp_2); // Default configuration, sending padding on the first sending module. const size_t requested_padding_bytes = 1000; const size_t sent_padding_bytes = 890; EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)) .Times(1) .WillOnce(Return(sent_padding_bytes)); EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0); EXPECT_EQ(sent_padding_bytes, packet_router_->TimeToSendPadding(requested_padding_bytes)); // Let only the second module be sending and verify the padding request is // routed there. EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false)); EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0); EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_2, TimeToSendPadding(_)) .Times(1) .WillOnce(Return(sent_padding_bytes)); EXPECT_EQ(sent_padding_bytes, packet_router_->TimeToSendPadding(requested_padding_bytes)); // No sending module at all. EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false)); EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0); EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false)); EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0); EXPECT_EQ(static_cast(0), packet_router_->TimeToSendPadding(requested_padding_bytes)); packet_router_->RemoveRtpModule(&rtp_1); // rtp_1 has been removed, try sending padding and make sure rtp_1 isn't asked // to send by not expecting any calls. Instead verify rtp_2 is called. EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes)).Times(1); EXPECT_EQ(static_cast(0), packet_router_->TimeToSendPadding(requested_padding_bytes)); packet_router_->RemoveRtpModule(&rtp_2); } } // namespace webrtc