/* * 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 "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/video_engine/payload_router.h" using ::testing::_; using ::testing::AnyNumber; using ::testing::NiceMock; using ::testing::Return; namespace webrtc { class PayloadRouterTest : public ::testing::Test { protected: virtual void SetUp() { payload_router_.reset(new PayloadRouter()); } scoped_ptr payload_router_; }; TEST_F(PayloadRouterTest, SendOnOneModule) { MockRtpRtcp rtp; std::list modules(1, &rtp); payload_router_->SetSendingRtpModules(modules); uint8_t payload = 'a'; FrameType frame_type = kVideoFrameKey; int8_t payload_type = 96; EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, NULL)) .Times(0); EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0, &payload, 1, NULL, NULL)); payload_router_->set_active(true); EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, NULL)) .Times(1); EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0, &payload, 1, NULL, NULL)); payload_router_->set_active(false); EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, NULL)) .Times(0); EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0, &payload, 1, NULL, NULL)); payload_router_->set_active(true); EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, NULL)) .Times(1); EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0, &payload, 1, NULL, NULL)); modules.clear(); payload_router_->SetSendingRtpModules(modules); EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, NULL)) .Times(0); EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0, &payload, 1, NULL, NULL)); } TEST_F(PayloadRouterTest, SendSimulcast) { MockRtpRtcp rtp_1; MockRtpRtcp rtp_2; std::list modules; modules.push_back(&rtp_1); modules.push_back(&rtp_2); payload_router_->SetSendingRtpModules(modules); uint8_t payload_1 = 'a'; FrameType frame_type_1 = kVideoFrameKey; int8_t payload_type_1 = 96; RTPVideoHeader rtp_hdr_1; rtp_hdr_1.simulcastIdx = 0; payload_router_->set_active(true); EXPECT_CALL(rtp_1, SendOutgoingData(frame_type_1, payload_type_1, 0, 0, _, 1, NULL, &rtp_hdr_1)) .Times(1); EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _)) .Times(0); EXPECT_TRUE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0, &payload_1, 1, NULL, &rtp_hdr_1)); uint8_t payload_2 = 'b'; FrameType frame_type_2 = kVideoFrameDelta; int8_t payload_type_2 = 97; RTPVideoHeader rtp_hdr_2; rtp_hdr_2.simulcastIdx = 1; EXPECT_CALL(rtp_2, SendOutgoingData(frame_type_2, payload_type_2, 0, 0, _, 1, NULL, &rtp_hdr_2)) .Times(1); EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _)) .Times(0); EXPECT_TRUE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0, &payload_2, 1, NULL, &rtp_hdr_2)); payload_router_->set_active(false); EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _)) .Times(0); EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _)) .Times(0); EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0, &payload_1, 1, NULL, &rtp_hdr_1)); EXPECT_FALSE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0, &payload_2, 1, NULL, &rtp_hdr_2)); } TEST_F(PayloadRouterTest, MaxPayloadLength) { // Without any limitations from the modules, verify we get the max payload // length for IP/UDP/SRTP with a MTU of 150 bytes. const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4; EXPECT_EQ(kDefaultMaxLength, payload_router_->DefaultMaxPayloadLength()); EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength()); MockRtpRtcp rtp_1; MockRtpRtcp rtp_2; std::list modules; modules.push_back(&rtp_1); modules.push_back(&rtp_2); payload_router_->SetSendingRtpModules(modules); // Modules return a higher length than the default value. EXPECT_CALL(rtp_1, MaxDataPayloadLength()) .Times(1) .WillOnce(Return(kDefaultMaxLength + 10)); EXPECT_CALL(rtp_2, MaxDataPayloadLength()) .Times(1) .WillOnce(Return(kDefaultMaxLength + 10)); EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength()); // The modules return a value lower than default. const size_t kTestMinPayloadLength = 1001; EXPECT_CALL(rtp_1, MaxDataPayloadLength()) .Times(1) .WillOnce(Return(kTestMinPayloadLength + 10)); EXPECT_CALL(rtp_2, MaxDataPayloadLength()) .Times(1) .WillOnce(Return(kTestMinPayloadLength)); EXPECT_EQ(kTestMinPayloadLength, payload_router_->MaxPayloadLength()); } TEST_F(PayloadRouterTest, TimeToSendPacket) { MockRtpRtcp rtp_1; MockRtpRtcp rtp_2; std::list modules; modules.push_back(&rtp_1); modules.push_back(&rtp_2); payload_router_->SetSendingRtpModules(modules); 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(payload_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(payload_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(payload_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(payload_router_->TimeToSendPacket( kSsrc1 + kSsrc2, sequence_number, timestamp, retransmission)); } TEST_F(PayloadRouterTest, TimeToSendPadding) { MockRtpRtcp rtp_1; MockRtpRtcp rtp_2; std::list modules; modules.push_back(&rtp_1); modules.push_back(&rtp_2); payload_router_->SetSendingRtpModules(modules); // 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, payload_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, payload_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), payload_router_->TimeToSendPadding(requested_padding_bytes)); } } // namespace webrtc