/* * 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/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/video/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()); } rtc::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)); // Inactive. 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)); // Invalid simulcast index. payload_router_->set_active(true); EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _)) .Times(0); EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _)) .Times(0); rtp_hdr_1.simulcastIdx = 2; EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0, &payload_1, 1, NULL, &rtp_hdr_1)); } 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, SetTargetSendBitrates) { MockRtpRtcp rtp_1; MockRtpRtcp rtp_2; std::list modules; modules.push_back(&rtp_1); modules.push_back(&rtp_2); payload_router_->SetSendingRtpModules(modules); const uint32_t bitrate_1 = 10000; const uint32_t bitrate_2 = 76543; std::vector bitrates(2, bitrate_1); bitrates[1] = bitrate_2; EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1)) .Times(1); EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2)) .Times(1); payload_router_->SetTargetSendBitrates(bitrates); bitrates.resize(1); EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1)) .Times(0); EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2)) .Times(0); payload_router_->SetTargetSendBitrates(bitrates); bitrates.resize(3); bitrates[1] = bitrate_2; bitrates[2] = bitrate_1 + bitrate_2; EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1)) .Times(1); EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2)) .Times(1); payload_router_->SetTargetSendBitrates(bitrates); } } // namespace webrtc