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/*
* Copyright (c) 2012 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 <algorithm>
#include <vector>
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
namespace webrtc {
namespace {
const uint64_t kTestPictureId = 12345678;
const uint8_t kSliPictureId = 156;
class RtcpCallback : public RtcpIntraFrameObserver {
public:
void SetModule(RtpRtcp* module) {
_rtpRtcpModule = module;
}
virtual void OnRTCPPacketTimeout(const int32_t id) {
}
virtual void OnLipSyncUpdate(const int32_t id,
const int32_t audioVideoOffset) {}
virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {}
virtual void OnReceivedSLI(uint32_t ssrc,
uint8_t pictureId) {
EXPECT_EQ(kSliPictureId & 0x3f, pictureId);
}
virtual void OnReceivedRPSI(uint32_t ssrc,
uint64_t pictureId) {
EXPECT_EQ(kTestPictureId, pictureId);
}
virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {}
private:
RtpRtcp* _rtpRtcpModule;
};
class TestRtpFeedback : public NullRtpFeedback {
public:
explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
virtual ~TestRtpFeedback() {}
void OnIncomingSSRCChanged(const uint32_t ssrc) override {
rtp_rtcp_->SetRemoteSSRC(ssrc);
}
private:
RtpRtcp* rtp_rtcp_;
};
class RtpRtcpRtcpTest : public ::testing::Test {
protected:
RtpRtcpRtcpTest() : fake_clock(123456) {
test_csrcs.push_back(1234);
test_csrcs.push_back(2345);
test_ssrc = 3456;
test_timestamp = 4567;
test_sequence_number = 2345;
}
~RtpRtcpRtcpTest() {}
virtual void SetUp() {
receiver = new TestRtpReceiver();
transport1 = new LoopBackTransport();
transport2 = new LoopBackTransport();
myRTCPFeedback1 = new RtcpCallback();
myRTCPFeedback2 = new RtcpCallback();
receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock));
receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
RtpRtcp::Configuration configuration;
configuration.audio = true;
configuration.clock = &fake_clock;
configuration.receive_statistics = receive_statistics1_.get();
configuration.outgoing_transport = transport1;
configuration.intra_frame_callback = myRTCPFeedback1;
rtp_payload_registry1_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
rtp_payload_registry2_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
module1 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_feedback1_.reset(new TestRtpFeedback(module1));
rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
&fake_clock, NULL, receiver, rtp_feedback1_.get(),
rtp_payload_registry1_.get()));
configuration.receive_statistics = receive_statistics2_.get();
configuration.outgoing_transport = transport2;
configuration.intra_frame_callback = myRTCPFeedback2;
module2 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_feedback2_.reset(new TestRtpFeedback(module2));
rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
&fake_clock, NULL, receiver, rtp_feedback2_.get(),
rtp_payload_registry2_.get()));
transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
rtp_receiver2_.get(), receive_statistics2_.get());
transport2->SetSendModule(module1, rtp_payload_registry1_.get(),
rtp_receiver1_.get(), receive_statistics1_.get());
myRTCPFeedback1->SetModule(module1);
myRTCPFeedback2->SetModule(module2);
module1->SetRTCPStatus(RtcpMode::kCompound);
module2->SetRTCPStatus(RtcpMode::kCompound);
module2->SetSSRC(test_ssrc + 1);
module1->SetSSRC(test_ssrc);
module1->SetSequenceNumber(test_sequence_number);
module1->SetStartTimestamp(test_timestamp);
module1->SetCsrcs(test_csrcs);
EXPECT_EQ(0, module1->SetCNAME("john.doe@test.test"));
EXPECT_EQ(0, module1->SetSendingStatus(true));
CodecInst voice_codec;
voice_codec.pltype = 96;
voice_codec.plfreq = 8000;
voice_codec.rate = 64000;
memcpy(voice_codec.plname, "PCMU", 5);
EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
voice_codec.plname,
voice_codec.pltype,
voice_codec.plfreq,
voice_codec.channels,
(voice_codec.rate < 0) ? 0 : voice_codec.rate));
EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
voice_codec.plname,
voice_codec.pltype,
voice_codec.plfreq,
voice_codec.channels,
(voice_codec.rate < 0) ? 0 : voice_codec.rate));
// We need to send one RTP packet to get the RTCP packet to be accepted by
// the receiving module.
// send RTP packet with the data "testtest"
const uint8_t test[9] = "testtest";
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
0, -1, test, 8));
}
virtual void TearDown() {
delete module1;
delete module2;
delete myRTCPFeedback1;
delete myRTCPFeedback2;
delete transport1;
delete transport2;
delete receiver;
}
rtc::scoped_ptr<TestRtpFeedback> rtp_feedback1_;
rtc::scoped_ptr<TestRtpFeedback> rtp_feedback2_;
rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
rtc::scoped_ptr<ReceiveStatistics> receive_statistics2_;
rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
rtc::scoped_ptr<RtpReceiver> rtp_receiver1_;
rtc::scoped_ptr<RtpReceiver> rtp_receiver2_;
RtpRtcp* module1;
RtpRtcp* module2;
TestRtpReceiver* receiver;
LoopBackTransport* transport1;
LoopBackTransport* transport2;
RtcpCallback* myRTCPFeedback1;
RtcpCallback* myRTCPFeedback2;
uint32_t test_ssrc;
uint32_t test_timestamp;
uint16_t test_sequence_number;
std::vector<uint32_t> test_csrcs;
SimulatedClock fake_clock;
};
TEST_F(RtpRtcpRtcpTest, RTCP_PLI_RPSI) {
EXPECT_EQ(0, module1->SendRTCPReferencePictureSelection(kTestPictureId));
EXPECT_EQ(0, module1->SendRTCPSliceLossIndication(kSliPictureId));
}
TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
uint32_t testOfCSRC[webrtc::kRtpCsrcSize];
EXPECT_EQ(2, rtp_receiver2_->CSRCs(testOfCSRC));
EXPECT_EQ(test_csrcs[0], testOfCSRC[0]);
EXPECT_EQ(test_csrcs[1], testOfCSRC[1]);
// Set cname of mixed.
EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[0], "john@192.168.0.1"));
EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[1], "jane@192.168.0.2"));
EXPECT_EQ(-1, module1->RemoveMixedCNAME(test_csrcs[0] + 1));
EXPECT_EQ(0, module1->RemoveMixedCNAME(test_csrcs[1]));
EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[1], "jane@192.168.0.2"));
// send RTCP packet, triggered by timer
fake_clock.AdvanceTimeMilliseconds(7500);
module1->Process();
fake_clock.AdvanceTimeMilliseconds(100);
module2->Process();
char cName[RTCP_CNAME_SIZE];
EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC() + 1, cName));
// Check multiple CNAME.
EXPECT_EQ(0, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
EXPECT_EQ(0, strncmp(cName, "john.doe@test.test", RTCP_CNAME_SIZE));
EXPECT_EQ(0, module2->RemoteCNAME(test_csrcs[0], cName));
EXPECT_EQ(0, strncmp(cName, "john@192.168.0.1", RTCP_CNAME_SIZE));
EXPECT_EQ(0, module2->RemoteCNAME(test_csrcs[1], cName));
EXPECT_EQ(0, strncmp(cName, "jane@192.168.0.2", RTCP_CNAME_SIZE));
EXPECT_EQ(0, module1->SetSendingStatus(false));
// Test that BYE clears the CNAME
EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
}
TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
std::vector<RTCPReportBlock> report_blocks;
EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
EXPECT_EQ(0u, report_blocks.size());
// send RTCP packet, triggered by timer
fake_clock.AdvanceTimeMilliseconds(7500);
module1->Process();
fake_clock.AdvanceTimeMilliseconds(100);
module2->Process();
EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
ASSERT_EQ(1u, report_blocks.size());
// |test_ssrc+1| is the SSRC of module2 that send the report.
EXPECT_EQ(test_ssrc+1, report_blocks[0].remoteSSRC);
EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC);
EXPECT_EQ(0u, report_blocks[0].cumulativeLost);
EXPECT_LT(0u, report_blocks[0].delaySinceLastSR);
EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
EXPECT_EQ(0u, report_blocks[0].fractionLost);
}
} // namespace
} // namespace webrtc
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