50e28166af
This cl just moves the logic form the default module SetTargetSendBitrates to PayloadRouter. There might be glitch / mismatch in size between trate the vector and rtp modules. This was the same in the default module and is quite hard to protect from before we have the new video API. I also removed some test form rtp_rtcp_impl_unittest that were affected by this change. The test tests code that isn't implemented, hence the DISABLED_, and this will never be implemented in the RTP module, rather the payload router in the future. BUG=769 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42419004 Cr-Commit-Position: refs/heads/master@{#8453} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8453 4adac7df-926f-26a2-2b94-8c16560cd09d
351 lines
12 KiB
C++
351 lines
12 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 "testing/gmock/include/gmock/gmock.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/video_engine/payload_router.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 PayloadRouterTest : public ::testing::Test {
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protected:
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virtual void SetUp() {
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payload_router_.reset(new PayloadRouter());
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}
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scoped_ptr<PayloadRouter> payload_router_;
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};
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TEST_F(PayloadRouterTest, SendOnOneModule) {
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MockRtpRtcp rtp;
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std::list<RtpRtcp*> modules(1, &rtp);
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payload_router_->SetSendingRtpModules(modules);
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uint8_t payload = 'a';
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FrameType frame_type = kVideoFrameKey;
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int8_t payload_type = 96;
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(0);
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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payload_router_->set_active(true);
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(1);
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EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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payload_router_->set_active(false);
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(0);
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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payload_router_->set_active(true);
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(1);
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EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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modules.clear();
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payload_router_->SetSendingRtpModules(modules);
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(0);
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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}
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TEST_F(PayloadRouterTest, SendSimulcast) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::list<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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payload_router_->SetSendingRtpModules(modules);
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uint8_t payload_1 = 'a';
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FrameType frame_type_1 = kVideoFrameKey;
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int8_t payload_type_1 = 96;
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RTPVideoHeader rtp_hdr_1;
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rtp_hdr_1.simulcastIdx = 0;
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payload_router_->set_active(true);
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EXPECT_CALL(rtp_1, SendOutgoingData(frame_type_1, payload_type_1, 0, 0, _, 1,
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NULL, &rtp_hdr_1))
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.Times(1);
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EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_TRUE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
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&payload_1, 1, NULL, &rtp_hdr_1));
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uint8_t payload_2 = 'b';
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FrameType frame_type_2 = kVideoFrameDelta;
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int8_t payload_type_2 = 97;
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RTPVideoHeader rtp_hdr_2;
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rtp_hdr_2.simulcastIdx = 1;
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EXPECT_CALL(rtp_2, SendOutgoingData(frame_type_2, payload_type_2, 0, 0, _, 1,
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NULL, &rtp_hdr_2))
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.Times(1);
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EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_TRUE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
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&payload_2, 1, NULL, &rtp_hdr_2));
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// Inactive.
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payload_router_->set_active(false);
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EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
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&payload_1, 1, NULL, &rtp_hdr_1));
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
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&payload_2, 1, NULL, &rtp_hdr_2));
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// Invalid simulcast index.
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payload_router_->set_active(true);
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EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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rtp_hdr_1.simulcastIdx = 2;
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
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&payload_1, 1, NULL, &rtp_hdr_1));
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}
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TEST_F(PayloadRouterTest, MaxPayloadLength) {
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// Without any limitations from the modules, verify we get the max payload
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// length for IP/UDP/SRTP with a MTU of 150 bytes.
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const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4;
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EXPECT_EQ(kDefaultMaxLength, payload_router_->DefaultMaxPayloadLength());
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EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::list<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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payload_router_->SetSendingRtpModules(modules);
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// Modules return a higher length than the default value.
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EXPECT_CALL(rtp_1, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kDefaultMaxLength + 10));
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EXPECT_CALL(rtp_2, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kDefaultMaxLength + 10));
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EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
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// The modules return a value lower than default.
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const size_t kTestMinPayloadLength = 1001;
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EXPECT_CALL(rtp_1, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kTestMinPayloadLength + 10));
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EXPECT_CALL(rtp_2, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kTestMinPayloadLength));
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EXPECT_EQ(kTestMinPayloadLength, payload_router_->MaxPayloadLength());
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}
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TEST_F(PayloadRouterTest, TimeToSendPacket) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::list<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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payload_router_->SetSendingRtpModules(modules);
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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())
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.Times(1)
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.WillOnce(Return(true));
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EXPECT_CALL(rtp_1, SSRC())
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.Times(1)
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.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(_, _, _, _))
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.Times(0);
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EXPECT_TRUE(payload_router_->TimeToSendPacket(
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kSsrc1, sequence_number, 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())
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.Times(1)
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.WillOnce(Return(false));
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EXPECT_CALL(rtp_2, SendingMedia())
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.Times(1)
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.WillOnce(Return(true));
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EXPECT_CALL(rtp_2, SSRC())
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.Times(1)
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.WillOnce(Return(kSsrc2));
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EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _))
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.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(payload_router_->TimeToSendPacket(
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kSsrc2, sequence_number, 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())
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.Times(1)
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.WillOnce(Return(false));
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EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _,_))
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.Times(0);
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EXPECT_CALL(rtp_2, SendingMedia())
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.Times(1)
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.WillOnce(Return(false));
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _))
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.Times(0);
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EXPECT_TRUE(payload_router_->TimeToSendPacket(
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kSsrc1, sequence_number, 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())
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.Times(1)
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.WillOnce(Return(true));
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EXPECT_CALL(rtp_1, SSRC())
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.Times(1)
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.WillOnce(Return(kSsrc1));
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EXPECT_CALL(rtp_2, SendingMedia())
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.Times(1)
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.WillOnce(Return(true));
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EXPECT_CALL(rtp_2, SSRC())
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.Times(1)
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.WillOnce(Return(kSsrc2));
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EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _,_))
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.Times(0);
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _))
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.Times(0);
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EXPECT_TRUE(payload_router_->TimeToSendPacket(
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kSsrc1 + kSsrc2, sequence_number, timestamp, retransmission));
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}
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TEST_F(PayloadRouterTest, TimeToSendPadding) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::list<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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payload_router_->SetSendingRtpModules(modules);
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// Default configuration, sending padding on the first sending module.
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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())
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.Times(1)
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.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, TimeToSendPadding(_))
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.Times(0);
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EXPECT_EQ(sent_padding_bytes,
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payload_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())
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.Times(1)
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.WillOnce(Return(false));
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EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
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.Times(0);
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EXPECT_CALL(rtp_2, SendingMedia())
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.Times(1)
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.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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payload_router_->TimeToSendPadding(requested_padding_bytes));
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// No sending module at all.
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EXPECT_CALL(rtp_1, SendingMedia())
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.Times(1)
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.WillOnce(Return(false));
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EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
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.Times(0);
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EXPECT_CALL(rtp_2, SendingMedia())
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.Times(1)
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.WillOnce(Return(false));
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EXPECT_CALL(rtp_2, TimeToSendPadding(_))
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.Times(0);
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EXPECT_EQ(static_cast<size_t>(0),
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payload_router_->TimeToSendPadding(requested_padding_bytes));
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}
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TEST_F(PayloadRouterTest, SetTargetSendBitrates) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::list<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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payload_router_->SetSendingRtpModules(modules);
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const uint32_t bitrate_1 = 10000;
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const uint32_t bitrate_2 = 76543;
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std::vector<uint32_t> bitrates (2, bitrate_1);
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bitrates[1] = bitrate_2;
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EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
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.Times(1);
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EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
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.Times(1);
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payload_router_->SetTargetSendBitrates(bitrates);
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bitrates.resize(1);
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EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
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.Times(0);
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EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
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.Times(0);
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payload_router_->SetTargetSendBitrates(bitrates);
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bitrates.resize(3);
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bitrates[1] = bitrate_2;
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bitrates[2] = bitrate_1 + bitrate_2;
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EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
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.Times(1);
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EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
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.Times(1);
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payload_router_->SetTargetSendBitrates(bitrates);
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}
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} // namespace webrtc
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