
BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8098 4adac7df-926f-26a2-2b94-8c16560cd09d
191 lines
6.1 KiB
C++
191 lines
6.1 KiB
C++
/*
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* Copyright (c) 2012 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 "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
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#include <algorithm>
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#include <vector>
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using namespace webrtc;
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namespace webrtc {
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void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
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RTPPayloadRegistry* payload_registry,
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RtpReceiver* receiver,
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ReceiveStatistics* receive_statistics) {
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rtp_rtcp_module_ = rtp_rtcp_module;
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rtp_payload_registry_ = payload_registry;
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rtp_receiver_ = receiver;
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receive_statistics_ = receive_statistics;
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}
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void LoopBackTransport::DropEveryNthPacket(int n) {
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packet_loss_ = n;
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}
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int LoopBackTransport::SendPacket(int channel, const void* data, size_t len) {
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count_++;
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if (packet_loss_ > 0) {
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if ((count_ % packet_loss_) == 0) {
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return len;
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}
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}
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RTPHeader header;
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scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
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if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
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return -1;
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}
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PayloadUnion payload_specific;
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if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
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&payload_specific)) {
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return -1;
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}
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receive_statistics_->IncomingPacket(header, len, false);
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if (!rtp_receiver_->IncomingRtpPacket(header,
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static_cast<const uint8_t*>(data), len,
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payload_specific, true)) {
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return -1;
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}
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return len;
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}
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int LoopBackTransport::SendRTCPPacket(int channel,
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const void* data,
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size_t len) {
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if (rtp_rtcp_module_->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
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return -1;
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}
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return static_cast<int>(len);
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}
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int32_t TestRtpReceiver::OnReceivedPayloadData(
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const uint8_t* payload_data,
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const size_t payload_size,
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const webrtc::WebRtcRTPHeader* rtp_header) {
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EXPECT_LE(payload_size, sizeof(payload_data_));
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memcpy(payload_data_, payload_data, payload_size);
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memcpy(&rtp_header_, rtp_header, sizeof(rtp_header_));
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payload_size_ = payload_size;
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return 0;
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}
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} // namespace webrtc
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class RtpRtcpAPITest : public ::testing::Test {
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protected:
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RtpRtcpAPITest() : fake_clock_(123456) {
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test_csrcs_.push_back(1234);
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test_csrcs_.push_back(2345);
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test_id = 123;
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test_ssrc_ = 3456;
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test_timestamp_ = 4567;
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test_sequence_number_ = 2345;
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}
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~RtpRtcpAPITest() {}
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virtual void SetUp() OVERRIDE {
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RtpRtcp::Configuration configuration;
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configuration.id = test_id;
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configuration.audio = true;
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configuration.clock = &fake_clock_;
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module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
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rtp_payload_registry_.reset(new RTPPayloadRegistry(
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RTPPayloadStrategy::CreateStrategy(true)));
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rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
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test_id, &fake_clock_, NULL, NULL, NULL, rtp_payload_registry_.get()));
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}
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int test_id;
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scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
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scoped_ptr<RtpReceiver> rtp_receiver_;
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scoped_ptr<RtpRtcp> module_;
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uint32_t test_ssrc_;
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uint32_t test_timestamp_;
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uint16_t test_sequence_number_;
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std::vector<uint32_t> test_csrcs_;
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SimulatedClock fake_clock_;
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};
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TEST_F(RtpRtcpAPITest, Basic) {
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module_->SetSequenceNumber(test_sequence_number_);
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EXPECT_EQ(test_sequence_number_, module_->SequenceNumber());
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module_->SetStartTimestamp(test_timestamp_);
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EXPECT_EQ(test_timestamp_, module_->StartTimestamp());
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EXPECT_FALSE(module_->Sending());
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EXPECT_EQ(0, module_->SetSendingStatus(true));
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EXPECT_TRUE(module_->Sending());
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}
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TEST_F(RtpRtcpAPITest, MTU) {
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EXPECT_EQ(-1, module_->SetMaxTransferUnit(10));
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EXPECT_EQ(-1, module_->SetMaxTransferUnit(IP_PACKET_SIZE + 1));
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EXPECT_EQ(0, module_->SetMaxTransferUnit(1234));
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EXPECT_EQ(1234 - 20 - 8, module_->MaxPayloadLength());
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EXPECT_EQ(0, module_->SetTransportOverhead(true, true, 12));
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EXPECT_EQ(1234 - 20 - 20 - 20 - 12, module_->MaxPayloadLength());
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EXPECT_EQ(0, module_->SetTransportOverhead(false, false, 0));
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EXPECT_EQ(1234 - 20 - 8, module_->MaxPayloadLength());
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}
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TEST_F(RtpRtcpAPITest, SSRC) {
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module_->SetSSRC(test_ssrc_);
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EXPECT_EQ(test_ssrc_, module_->SSRC());
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}
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TEST_F(RtpRtcpAPITest, RTCP) {
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EXPECT_EQ(kRtcpOff, module_->RTCP());
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module_->SetRTCPStatus(kRtcpCompound);
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EXPECT_EQ(kRtcpCompound, module_->RTCP());
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EXPECT_EQ(0, module_->SetCNAME("john.doe@test.test"));
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EXPECT_FALSE(module_->TMMBR());
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module_->SetTMMBRStatus(true);
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EXPECT_TRUE(module_->TMMBR());
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module_->SetTMMBRStatus(false);
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EXPECT_FALSE(module_->TMMBR());
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EXPECT_EQ(kNackOff, rtp_receiver_->NACK());
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rtp_receiver_->SetNACKStatus(kNackRtcp);
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EXPECT_EQ(kNackRtcp, rtp_receiver_->NACK());
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}
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TEST_F(RtpRtcpAPITest, RtxSender) {
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module_->SetRtxSendStatus(kRtxRetransmitted);
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EXPECT_EQ(kRtxRetransmitted, module_->RtxSendStatus());
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module_->SetRtxSendStatus(kRtxOff);
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EXPECT_EQ(kRtxOff, module_->RtxSendStatus());
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module_->SetRtxSendStatus(kRtxRetransmitted);
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EXPECT_EQ(kRtxRetransmitted, module_->RtxSendStatus());
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}
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TEST_F(RtpRtcpAPITest, RtxReceiver) {
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const uint32_t kRtxSsrc = 1;
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const int kRtxPayloadType = 119;
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EXPECT_FALSE(rtp_payload_registry_->RtxEnabled());
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rtp_payload_registry_->SetRtxSsrc(kRtxSsrc);
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rtp_payload_registry_->SetRtxPayloadType(kRtxPayloadType);
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EXPECT_TRUE(rtp_payload_registry_->RtxEnabled());
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RTPHeader rtx_header;
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rtx_header.ssrc = kRtxSsrc;
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rtx_header.payloadType = kRtxPayloadType;
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EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
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rtx_header.ssrc = 0;
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EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
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rtx_header.ssrc = kRtxSsrc;
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rtx_header.payloadType = 0;
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EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
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}
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