Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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Mirko Bonadei
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249
modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
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249
modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
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/*
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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 <algorithm>
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#include <memory>
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#include <vector>
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#include "webrtc/common_types.h"
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#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
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#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
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#include "webrtc/rtc_base/rate_limiter.h"
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#include "webrtc/test/gmock.h"
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#include "webrtc/test/gtest.h"
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namespace webrtc {
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namespace {
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class RtcpCallback : public RtcpIntraFrameObserver {
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public:
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void SetModule(RtpRtcp* module) {
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_rtpRtcpModule = module;
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}
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virtual void OnRTCPPacketTimeout(const int32_t id) {
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}
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virtual void OnLipSyncUpdate(const int32_t id,
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const int32_t audioVideoOffset) {}
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virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {}
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private:
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RtpRtcp* _rtpRtcpModule;
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};
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class TestRtpFeedback : public NullRtpFeedback {
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public:
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explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
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virtual ~TestRtpFeedback() {}
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void OnIncomingSSRCChanged(uint32_t ssrc) override {
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rtp_rtcp_->SetRemoteSSRC(ssrc);
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}
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private:
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RtpRtcp* rtp_rtcp_;
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};
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class RtpRtcpRtcpTest : public ::testing::Test {
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protected:
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RtpRtcpRtcpTest()
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: fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
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test_csrcs.push_back(1234);
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test_csrcs.push_back(2345);
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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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~RtpRtcpRtcpTest() {}
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virtual void SetUp() {
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receiver = new TestRtpReceiver();
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transport1 = new LoopBackTransport();
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transport2 = new LoopBackTransport();
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myRTCPFeedback1 = new RtcpCallback();
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myRTCPFeedback2 = new RtcpCallback();
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receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock));
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receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
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RtpRtcp::Configuration configuration;
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configuration.audio = true;
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configuration.clock = &fake_clock;
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configuration.receive_statistics = receive_statistics1_.get();
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configuration.outgoing_transport = transport1;
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configuration.intra_frame_callback = myRTCPFeedback1;
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configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
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rtp_payload_registry1_.reset(new RTPPayloadRegistry());
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rtp_payload_registry2_.reset(new RTPPayloadRegistry());
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module1 = RtpRtcp::CreateRtpRtcp(configuration);
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rtp_feedback1_.reset(new TestRtpFeedback(module1));
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rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
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&fake_clock, receiver, rtp_feedback1_.get(),
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rtp_payload_registry1_.get()));
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configuration.receive_statistics = receive_statistics2_.get();
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configuration.outgoing_transport = transport2;
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configuration.intra_frame_callback = myRTCPFeedback2;
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module2 = RtpRtcp::CreateRtpRtcp(configuration);
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rtp_feedback2_.reset(new TestRtpFeedback(module2));
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rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
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&fake_clock, receiver, rtp_feedback2_.get(),
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rtp_payload_registry2_.get()));
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transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
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rtp_receiver2_.get(), receive_statistics2_.get());
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transport2->SetSendModule(module1, rtp_payload_registry1_.get(),
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rtp_receiver1_.get(), receive_statistics1_.get());
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myRTCPFeedback1->SetModule(module1);
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myRTCPFeedback2->SetModule(module2);
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module1->SetRTCPStatus(RtcpMode::kCompound);
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module2->SetRTCPStatus(RtcpMode::kCompound);
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module2->SetSSRC(test_ssrc + 1);
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module1->SetSSRC(test_ssrc);
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module1->SetSequenceNumber(test_sequence_number);
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module1->SetStartTimestamp(test_timestamp);
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module1->SetCsrcs(test_csrcs);
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EXPECT_EQ(0, module1->SetCNAME("john.doe@test.test"));
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EXPECT_EQ(0, module1->SetSendingStatus(true));
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CodecInst voice_codec;
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voice_codec.pltype = 96;
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voice_codec.plfreq = 8000;
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voice_codec.rate = 64000;
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memcpy(voice_codec.plname, "PCMU", 5);
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EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
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EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(voice_codec));
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EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
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EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(voice_codec));
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// We need to send one RTP packet to get the RTCP packet to be accepted by
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// the receiving module.
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// send RTP packet with the data "testtest"
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const uint8_t test[9] = "testtest";
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EXPECT_EQ(true,
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module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
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test, 8, nullptr, nullptr, nullptr));
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}
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virtual void TearDown() {
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delete module1;
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delete module2;
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delete myRTCPFeedback1;
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delete myRTCPFeedback2;
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delete transport1;
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delete transport2;
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delete receiver;
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}
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std::unique_ptr<TestRtpFeedback> rtp_feedback1_;
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std::unique_ptr<TestRtpFeedback> rtp_feedback2_;
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std::unique_ptr<ReceiveStatistics> receive_statistics1_;
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std::unique_ptr<ReceiveStatistics> receive_statistics2_;
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std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
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std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
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std::unique_ptr<RtpReceiver> rtp_receiver1_;
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std::unique_ptr<RtpReceiver> rtp_receiver2_;
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RtpRtcp* module1;
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RtpRtcp* module2;
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TestRtpReceiver* receiver;
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LoopBackTransport* transport1;
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LoopBackTransport* transport2;
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RtcpCallback* myRTCPFeedback1;
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RtcpCallback* myRTCPFeedback2;
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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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RateLimiter retransmission_rate_limiter_;
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};
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TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
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uint32_t testOfCSRC[webrtc::kRtpCsrcSize];
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EXPECT_EQ(2, rtp_receiver2_->CSRCs(testOfCSRC));
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EXPECT_EQ(test_csrcs[0], testOfCSRC[0]);
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EXPECT_EQ(test_csrcs[1], testOfCSRC[1]);
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// Set cname of mixed.
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EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[0], "john@192.168.0.1"));
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EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[1], "jane@192.168.0.2"));
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EXPECT_EQ(-1, module1->RemoveMixedCNAME(test_csrcs[0] + 1));
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EXPECT_EQ(0, module1->RemoveMixedCNAME(test_csrcs[1]));
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EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[1], "jane@192.168.0.2"));
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// send RTCP packet, triggered by timer
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fake_clock.AdvanceTimeMilliseconds(7500);
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module1->Process();
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fake_clock.AdvanceTimeMilliseconds(100);
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module2->Process();
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char cName[RTCP_CNAME_SIZE];
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EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC() + 1, cName));
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// Check multiple CNAME.
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EXPECT_EQ(0, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
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EXPECT_EQ(0, strncmp(cName, "john.doe@test.test", RTCP_CNAME_SIZE));
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EXPECT_EQ(0, module2->RemoteCNAME(test_csrcs[0], cName));
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EXPECT_EQ(0, strncmp(cName, "john@192.168.0.1", RTCP_CNAME_SIZE));
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EXPECT_EQ(0, module2->RemoteCNAME(test_csrcs[1], cName));
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EXPECT_EQ(0, strncmp(cName, "jane@192.168.0.2", RTCP_CNAME_SIZE));
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EXPECT_EQ(0, module1->SetSendingStatus(false));
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// Test that BYE clears the CNAME
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EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
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}
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TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
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std::vector<RTCPReportBlock> report_blocks;
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EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
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EXPECT_EQ(0u, report_blocks.size());
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// send RTCP packet, triggered by timer
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fake_clock.AdvanceTimeMilliseconds(7500);
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module1->Process();
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fake_clock.AdvanceTimeMilliseconds(100);
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module2->Process();
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EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
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ASSERT_EQ(1u, report_blocks.size());
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// |test_ssrc+1| is the SSRC of module2 that send the report.
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EXPECT_EQ(test_ssrc + 1, report_blocks[0].sender_ssrc);
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EXPECT_EQ(test_ssrc, report_blocks[0].source_ssrc);
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EXPECT_EQ(0u, report_blocks[0].packets_lost);
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EXPECT_LT(0u, report_blocks[0].delay_since_last_sender_report);
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EXPECT_EQ(test_sequence_number,
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report_blocks[0].extended_highest_sequence_number);
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EXPECT_EQ(0u, report_blocks[0].fraction_lost);
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}
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} // namespace
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} // namespace webrtc
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