Change some pointers to std::unique_ptr in rtp_rtcp tests.
Bug: none
Change-Id: Ia4e69e44bbda7b5b633b8be1779d105649f44930
Reviewed-on: https://webrtc-review.googlesource.com/94844
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24419}diff --git a/modules/rtp_rtcp/test/testAPI/test_api.cc b/modules/rtp_rtcp/test/testAPI/test_api.cc
index 5fa1de0..2fd4464 100644
--- a/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -13,7 +13,6 @@
#include <algorithm>
#include <memory>
#include <string>
-#include <vector>
#include "rtc_base/checks.h"
#include "rtc_base/rate_limiter.h"
@@ -96,10 +95,10 @@
module_->SetSSRC(kInitialSsrc);
}
- std::unique_ptr<RtpRtcp> module_;
SimulatedClock fake_clock_;
test::NullTransport null_transport_;
RateLimiter retransmission_rate_limiter_;
+ std::unique_ptr<RtpRtcp> module_;
};
TEST_F(RtpRtcpAPITest, Basic) {
diff --git a/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 2618433..fd7b8a7 100644
--- a/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -88,78 +88,79 @@
class RtpRtcpAudioTest : public ::testing::Test {
protected:
RtpRtcpAudioTest()
- : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {}
- ~RtpRtcpAudioTest() override = default;
-
- void SetUp() override {
- receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock_));
- receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock_));
-
+ : fake_clock_(123456),
+ retransmission_rate_limiter_(&fake_clock_, 1000),
+ receive_statistics1_(ReceiveStatistics::Create(&fake_clock_)),
+ receive_statistics2_(ReceiveStatistics::Create(&fake_clock_)),
+ rtp_receiver1_(
+ RtpReceiver::CreateAudioReceiver(&fake_clock_,
+ &data_receiver1_,
+ &rtp_payload_registry1_)),
+ rtp_receiver2_(
+ RtpReceiver::CreateAudioReceiver(&fake_clock_,
+ &data_receiver2_,
+ &rtp_payload_registry2_)) {
RtpRtcp::Configuration configuration;
configuration.audio = true;
configuration.clock = &fake_clock_;
configuration.receive_statistics = receive_statistics1_.get();
- configuration.outgoing_transport = &transport1;
+ configuration.outgoing_transport = &transport1_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
-
- module1.reset(RtpRtcp::CreateRtpRtcp(configuration));
- rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
- &fake_clock_, &data_receiver1, &rtp_payload_registry1_));
+ module1_.reset(RtpRtcp::CreateRtpRtcp(configuration));
configuration.receive_statistics = receive_statistics2_.get();
- configuration.outgoing_transport = &transport2;
+ configuration.outgoing_transport = &transport2_;
+ module2_.reset(RtpRtcp::CreateRtpRtcp(configuration));
- module2.reset(RtpRtcp::CreateRtpRtcp(configuration));
- rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
- &fake_clock_, &data_receiver2, &rtp_payload_registry2_));
-
- transport1.SetSendModule(module2.get(), &rtp_payload_registry2_,
- rtp_receiver2_.get(), receive_statistics2_.get());
- transport2.SetSendModule(module1.get(), &rtp_payload_registry1_,
- rtp_receiver1_.get(), receive_statistics1_.get());
+ transport1_.SetSendModule(module2_.get(), &rtp_payload_registry2_,
+ rtp_receiver2_.get(), receive_statistics2_.get());
+ transport2_.SetSendModule(module1_.get(), &rtp_payload_registry1_,
+ rtp_receiver1_.get(), receive_statistics1_.get());
}
+ ~RtpRtcpAudioTest() override = default;
+
void RegisterPayload(const CodecInst& codec) {
- EXPECT_EQ(0, module1->RegisterSendPayload(codec));
+ EXPECT_EQ(0, module1_->RegisterSendPayload(codec));
EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(codec.pltype,
CodecInstToSdp(codec)));
- EXPECT_EQ(0, module2->RegisterSendPayload(codec));
+ EXPECT_EQ(0, module2_->RegisterSendPayload(codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(codec.pltype,
CodecInstToSdp(codec)));
}
- VerifyingAudioReceiver data_receiver1;
- VerifyingAudioReceiver data_receiver2;
+ SimulatedClock fake_clock_;
+ RateLimiter retransmission_rate_limiter_;
+ VerifyingAudioReceiver data_receiver1_;
+ VerifyingAudioReceiver data_receiver2_;
std::unique_ptr<ReceiveStatistics> receive_statistics1_;
std::unique_ptr<ReceiveStatistics> receive_statistics2_;
RTPPayloadRegistry rtp_payload_registry1_;
RTPPayloadRegistry rtp_payload_registry2_;
std::unique_ptr<RtpReceiver> rtp_receiver1_;
std::unique_ptr<RtpReceiver> rtp_receiver2_;
- std::unique_ptr<RtpRtcp> module1;
- std::unique_ptr<RtpRtcp> module2;
- LoopBackTransport transport1;
- LoopBackTransport transport2;
- SimulatedClock fake_clock_;
- RateLimiter retransmission_rate_limiter_;
+ std::unique_ptr<RtpRtcp> module1_;
+ std::unique_ptr<RtpRtcp> module2_;
+ LoopBackTransport transport1_;
+ LoopBackTransport transport2_;
};
TEST_F(RtpRtcpAudioTest, Basic) {
- module1->SetSSRC(kSsrc);
- module1->SetStartTimestamp(kTimestamp);
+ module1_->SetSSRC(kSsrc);
+ module1_->SetStartTimestamp(kTimestamp);
// Test detection at the end of a DTMF tone.
- // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
+ // EXPECT_EQ(0, module2_->SetTelephoneEventForwardToDecoder(true));
- EXPECT_EQ(0, module1->SetSendingStatus(true));
+ EXPECT_EQ(0, module1_->SetSendingStatus(true));
// Start basic RTP test.
// Send an empty RTP packet.
// Should fail since we have not registered the payload type.
- EXPECT_FALSE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
- kPcmuPayloadType, 0, -1, nullptr, 0,
- nullptr, nullptr, nullptr));
+ EXPECT_FALSE(module1_->SendOutgoingData(webrtc::kAudioFrameSpeech,
+ kPcmuPayloadType, 0, -1, nullptr, 0,
+ nullptr, nullptr, nullptr));
CodecInst voice_codec = {};
voice_codec.pltype = kPcmuPayloadType;
@@ -168,9 +169,9 @@
memcpy(voice_codec.plname, "PCMU", 5);
RegisterPayload(voice_codec);
- EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
- kPcmuPayloadType, 0, -1, kTestPayload,
- 4, nullptr, nullptr, nullptr));
+ EXPECT_TRUE(module1_->SendOutgoingData(webrtc::kAudioFrameSpeech,
+ kPcmuPayloadType, 0, -1, kTestPayload,
+ 4, nullptr, nullptr, nullptr));
EXPECT_EQ(kSsrc, rtp_receiver2_->SSRC());
uint32_t timestamp;
@@ -189,16 +190,16 @@
memcpy(voice_codec.plname, "PCMU", 5);
RegisterPayload(voice_codec);
- module1->SetSSRC(kSsrc);
- module1->SetStartTimestamp(kTimestamp);
- EXPECT_EQ(0, module1->SetSendingStatus(true));
+ module1_->SetSSRC(kSsrc);
+ module1_->SetStartTimestamp(kTimestamp);
+ EXPECT_EQ(0, module1_->SetSendingStatus(true));
// Prepare for DTMF.
voice_codec.pltype = kDtmfPayloadType;
voice_codec.plfreq = 8000;
memcpy(voice_codec.plname, "telephone-event", 16);
- EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
+ EXPECT_EQ(0, module1_->RegisterSendPayload(voice_codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
voice_codec.pltype, CodecInstToSdp(voice_codec)));
@@ -207,37 +208,37 @@
// Send a DTMF tone using RFC 2833 (4733).
for (int i = 0; i < 16; i++) {
- EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10));
+ EXPECT_EQ(0, module1_->SendTelephoneEventOutband(i, timeStamp, 10));
}
timeStamp += 160; // Prepare for next packet.
// Send RTP packets for 16 tones a 160 ms 100ms
// pause between = 2560ms + 1600ms = 4160ms
for (; timeStamp <= 250 * 160; timeStamp += 160) {
- EXPECT_TRUE(module1->SendOutgoingData(
+ EXPECT_TRUE(module1_->SendOutgoingData(
webrtc::kAudioFrameSpeech, kPcmuPayloadType, timeStamp, -1,
kTestPayload, 4, nullptr, nullptr, nullptr));
fake_clock_.AdvanceTimeMilliseconds(20);
- module1->Process();
+ module1_->Process();
}
- EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
+ EXPECT_EQ(0, module1_->SendTelephoneEventOutband(32, 9000, 10));
for (; timeStamp <= 740 * 160; timeStamp += 160) {
- EXPECT_TRUE(module1->SendOutgoingData(
+ EXPECT_TRUE(module1_->SendOutgoingData(
webrtc::kAudioFrameSpeech, kPcmuPayloadType, timeStamp, -1,
kTestPayload, 4, nullptr, nullptr, nullptr));
fake_clock_.AdvanceTimeMilliseconds(20);
- module1->Process();
+ module1_->Process();
}
}
TEST_F(RtpRtcpAudioTest, ComfortNoise) {
- module1->SetSSRC(kSsrc);
- module1->SetStartTimestamp(kTimestamp);
+ module1_->SetSSRC(kSsrc);
+ module1_->SetStartTimestamp(kTimestamp);
- EXPECT_EQ(0, module1->SetSendingStatus(true));
+ EXPECT_EQ(0, module1_->SetSendingStatus(true));
- // Register PCMU and all four comfort noise codecs
+ // Register PCMU and all four comfort noise codecs.
CodecInst voice_codec = {};
voice_codec.pltype = kPcmuPayloadType;
voice_codec.plfreq = 8000;
@@ -258,7 +259,7 @@
for (const auto& c : kCngCodecs) {
uint32_t timestamp;
int64_t receive_time_ms;
- EXPECT_TRUE(module1->SendOutgoingData(
+ EXPECT_TRUE(module1_->SendOutgoingData(
webrtc::kAudioFrameSpeech, kPcmuPayloadType, in_timestamp, -1,
kTestPayload, 4, nullptr, nullptr, nullptr));
@@ -270,9 +271,9 @@
in_timestamp += 10;
fake_clock_.AdvanceTimeMilliseconds(20);
- EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameCN, c.payload_type,
- in_timestamp, -1, kTestPayload, 1,
- nullptr, nullptr, nullptr));
+ EXPECT_TRUE(module1_->SendOutgoingData(
+ webrtc::kAudioFrameCN, c.payload_type, in_timestamp, -1, kTestPayload,
+ 1, nullptr, nullptr, nullptr));
EXPECT_EQ(kSsrc, rtp_receiver2_->SSRC());
EXPECT_TRUE(
diff --git a/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index 7551e71..7d8628a 100644
--- a/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -20,7 +20,6 @@
#include "modules/rtp_rtcp/source/rtp_receiver_audio.h"
#include "modules/rtp_rtcp/test/testAPI/test_api.h"
#include "rtc_base/rate_limiter.h"
-#include "test/gmock.h"
#include "test/gtest.h"
namespace webrtc {
@@ -37,57 +36,53 @@
class RtpRtcpRtcpTest : public ::testing::Test {
protected:
RtpRtcpRtcpTest()
- : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {}
+ : fake_clock_(123456),
+ retransmission_rate_limiter_(&fake_clock_, 1000),
+ receive_statistics1_(ReceiveStatistics::Create(&fake_clock_)),
+ receive_statistics2_(ReceiveStatistics::Create(&fake_clock_)),
+ rtp_receiver1_(
+ RtpReceiver::CreateAudioReceiver(&fake_clock_,
+ &receiver_,
+ &rtp_payload_registry1_)),
+ rtp_receiver2_(
+ RtpReceiver::CreateAudioReceiver(&fake_clock_,
+ &receiver_,
+ &rtp_payload_registry2_)) {}
~RtpRtcpRtcpTest() override = default;
void SetUp() override {
- receiver = new TestRtpReceiver();
- transport1 = new LoopBackTransport();
- transport2 = new LoopBackTransport();
-
- 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.outgoing_transport = &transport1_;
configuration.intra_frame_callback = &rtcp_callback1_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
-
- module1 = RtpRtcp::CreateRtpRtcp(configuration);
-
- rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
- &fake_clock_, receiver, &rtp_payload_registry1_));
+ module1_.reset(RtpRtcp::CreateRtpRtcp(configuration));
configuration.receive_statistics = receive_statistics2_.get();
- configuration.outgoing_transport = transport2;
+ configuration.outgoing_transport = &transport2_;
configuration.intra_frame_callback = &rtcp_callback2_;
+ module2_.reset(RtpRtcp::CreateRtpRtcp(configuration));
- module2 = RtpRtcp::CreateRtpRtcp(configuration);
-
- rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
- &fake_clock_, receiver, &rtp_payload_registry2_));
-
- transport1->SetSendModule(module2, &rtp_payload_registry2_,
+ transport1_.SetSendModule(module2_.get(), &rtp_payload_registry2_,
rtp_receiver2_.get(), receive_statistics2_.get());
- transport2->SetSendModule(module1, &rtp_payload_registry1_,
+ transport2_.SetSendModule(module1_.get(), &rtp_payload_registry1_,
rtp_receiver1_.get(), receive_statistics1_.get());
- module1->SetRTCPStatus(RtcpMode::kCompound);
- module2->SetRTCPStatus(RtcpMode::kCompound);
+ module1_->SetRTCPStatus(RtcpMode::kCompound);
+ module2_->SetRTCPStatus(RtcpMode::kCompound);
- module2->SetSSRC(kSsrc + 1);
- module2->SetRemoteSSRC(kSsrc);
- module1->SetSSRC(kSsrc);
- module1->SetSequenceNumber(kSequenceNumber);
- module1->SetStartTimestamp(kTimestamp);
+ module2_->SetSSRC(kSsrc + 1);
+ module2_->SetRemoteSSRC(kSsrc);
+ module1_->SetSSRC(kSsrc);
+ module1_->SetSequenceNumber(kSequenceNumber);
+ module1_->SetStartTimestamp(kTimestamp);
- module1->SetCsrcs(kCsrcs);
- EXPECT_EQ(0, module1->SetCNAME("john.doe@test.test"));
+ module1_->SetCsrcs(kCsrcs);
+ EXPECT_EQ(0, module1_->SetCNAME("john.doe@test.test"));
- EXPECT_EQ(0, module1->SetSendingStatus(true));
+ EXPECT_EQ(0, module1_->SetSendingStatus(true));
CodecInst voice_codec;
voice_codec.pltype = 96;
@@ -95,10 +90,10 @@
voice_codec.rate = 64000;
memcpy(voice_codec.plname, "PCMU", 5);
- EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
+ EXPECT_EQ(0, module1_->RegisterSendPayload(voice_codec));
EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
voice_codec.pltype, CodecInstToSdp(voice_codec)));
- EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
+ EXPECT_EQ(0, module2_->RegisterSendPayload(voice_codec));
EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
voice_codec.pltype, CodecInstToSdp(voice_codec)));
@@ -107,84 +102,74 @@
// Send RTP packet with the data "testtest".
const uint8_t test[9] = "testtest";
EXPECT_EQ(true,
- module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
- test, 8, nullptr, nullptr, nullptr));
+ module1_->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
+ test, 8, nullptr, nullptr, nullptr));
}
- void TearDown() override {
- delete module1;
- delete module2;
- delete transport1;
- delete transport2;
- delete receiver;
- }
-
- RtcpCallback rtcp_callback1_;
- RtcpCallback rtcp_callback2_;
- RTPPayloadRegistry rtp_payload_registry1_;
- RTPPayloadRegistry rtp_payload_registry2_;
- std::unique_ptr<ReceiveStatistics> receive_statistics1_;
- std::unique_ptr<ReceiveStatistics> receive_statistics2_;
- std::unique_ptr<RtpReceiver> rtp_receiver1_;
- std::unique_ptr<RtpReceiver> rtp_receiver2_;
- RtpRtcp* module1;
- RtpRtcp* module2;
- TestRtpReceiver* receiver;
- LoopBackTransport* transport1;
- LoopBackTransport* transport2;
-
const std::vector<uint32_t> kCsrcs = {1234, 2345};
SimulatedClock fake_clock_;
RateLimiter retransmission_rate_limiter_;
+ RtcpCallback rtcp_callback1_;
+ RtcpCallback rtcp_callback2_;
+ RTPPayloadRegistry rtp_payload_registry1_;
+ RTPPayloadRegistry rtp_payload_registry2_;
+ TestRtpReceiver receiver_;
+ std::unique_ptr<ReceiveStatistics> receive_statistics1_;
+ std::unique_ptr<ReceiveStatistics> receive_statistics2_;
+ std::unique_ptr<RtpReceiver> rtp_receiver1_;
+ std::unique_ptr<RtpReceiver> rtp_receiver2_;
+ std::unique_ptr<RtpRtcp> module1_;
+ std::unique_ptr<RtpRtcp> module2_;
+ LoopBackTransport transport1_;
+ LoopBackTransport transport2_;
};
TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
// Set cname of mixed.
- EXPECT_EQ(0, module1->AddMixedCNAME(kCsrcs[0], "john@192.168.0.1"));
- EXPECT_EQ(0, module1->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2"));
+ EXPECT_EQ(0, module1_->AddMixedCNAME(kCsrcs[0], "john@192.168.0.1"));
+ EXPECT_EQ(0, module1_->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2"));
- EXPECT_EQ(-1, module1->RemoveMixedCNAME(kCsrcs[0] + 1));
- EXPECT_EQ(0, module1->RemoveMixedCNAME(kCsrcs[1]));
- EXPECT_EQ(0, module1->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2"));
+ EXPECT_EQ(-1, module1_->RemoveMixedCNAME(kCsrcs[0] + 1));
+ EXPECT_EQ(0, module1_->RemoveMixedCNAME(kCsrcs[1]));
+ EXPECT_EQ(0, module1_->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2"));
// Send RTCP packet, triggered by timer.
fake_clock_.AdvanceTimeMilliseconds(7500);
- module1->Process();
+ module1_->Process();
fake_clock_.AdvanceTimeMilliseconds(100);
- module2->Process();
+ module2_->Process();
char cName[RTCP_CNAME_SIZE];
- EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC() + 1, cName));
+ 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, module2_->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
EXPECT_EQ(0, strncmp(cName, "john.doe@test.test", RTCP_CNAME_SIZE));
- EXPECT_EQ(0, module2->RemoteCNAME(kCsrcs[0], cName));
+ EXPECT_EQ(0, module2_->RemoteCNAME(kCsrcs[0], cName));
EXPECT_EQ(0, strncmp(cName, "john@192.168.0.1", RTCP_CNAME_SIZE));
- EXPECT_EQ(0, module2->RemoteCNAME(kCsrcs[1], cName));
+ EXPECT_EQ(0, module2_->RemoteCNAME(kCsrcs[1], cName));
EXPECT_EQ(0, strncmp(cName, "jane@192.168.0.2", RTCP_CNAME_SIZE));
- EXPECT_EQ(0, module1->SetSendingStatus(false));
+ EXPECT_EQ(0, module1_->SetSendingStatus(false));
// Test that BYE clears the CNAME.
- EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), 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(0, module1_->RemoteRTCPStat(&report_blocks));
EXPECT_EQ(0u, report_blocks.size());
// Send RTCP packet, triggered by timer.
fake_clock_.AdvanceTimeMilliseconds(7500);
- module1->Process();
+ module1_->Process();
fake_clock_.AdvanceTimeMilliseconds(100);
- module2->Process();
+ module2_->Process();
- EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
+ EXPECT_EQ(0, module1_->RemoteRTCPStat(&report_blocks));
ASSERT_EQ(1u, report_blocks.size());
// |kSsrc+1| is the SSRC of module2 that send the report.
diff --git a/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index 4d8190b..aa524bd 100644
--- a/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -33,30 +33,30 @@
class RtpRtcpVideoTest : public ::testing::Test {
protected:
RtpRtcpVideoTest()
- : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {}
+ : fake_clock_(123456),
+ retransmission_rate_limiter_(&fake_clock_, 1000),
+ receive_statistics_(ReceiveStatistics::Create(&fake_clock_)),
+ rtp_receiver_(
+ RtpReceiver::CreateVideoReceiver(&fake_clock_,
+ &receiver_,
+ &rtp_payload_registry_)) {}
~RtpRtcpVideoTest() override = default;
void SetUp() override {
- transport_ = new LoopBackTransport();
- receiver_ = new TestRtpReceiver();
- receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock_));
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.clock = &fake_clock_;
- configuration.outgoing_transport = transport_;
+ configuration.outgoing_transport = &transport_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
-
- video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
- rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
- &fake_clock_, receiver_, &rtp_payload_registry_));
+ video_module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
video_module_->SetRTCPStatus(RtcpMode::kCompound);
video_module_->SetSSRC(kSsrc);
video_module_->SetStorePacketsStatus(true, 600);
EXPECT_EQ(0, video_module_->SetSendingStatus(true));
- transport_->SetSendModule(video_module_, &rtp_payload_registry_,
- rtp_receiver_.get(), receive_statistics_.get());
+ transport_.SetSendModule(video_module_.get(), &rtp_payload_registry_,
+ rtp_receiver_.get(), receive_statistics_.get());
VideoCodec video_codec;
memset(&video_codec, 0, sizeof(video_codec));
@@ -111,22 +111,16 @@
return padding_bytes_in_packet + header_length;
}
- void TearDown() override {
- delete video_module_;
- delete transport_;
- delete receiver_;
- }
-
- std::unique_ptr<ReceiveStatistics> receive_statistics_;
- RTPPayloadRegistry rtp_payload_registry_;
- std::unique_ptr<RtpReceiver> rtp_receiver_;
- RtpRtcp* video_module_;
- LoopBackTransport* transport_;
- TestRtpReceiver* receiver_;
uint8_t video_frame_[65000];
size_t payload_data_length_;
SimulatedClock fake_clock_;
RateLimiter retransmission_rate_limiter_;
+ std::unique_ptr<ReceiveStatistics> receive_statistics_;
+ RTPPayloadRegistry rtp_payload_registry_;
+ TestRtpReceiver receiver_;
+ std::unique_ptr<RtpReceiver> rtp_receiver_;
+ std::unique_ptr<RtpRtcp> video_module_;
+ LoopBackTransport transport_;
};
TEST_F(RtpRtcpVideoTest, BasicVideo) {
@@ -161,8 +155,8 @@
const size_t payload_length = packet_size - header.headerLength;
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
header, payload, payload_length, pl->typeSpecific));
- EXPECT_EQ(0u, receiver_->payload_size());
- EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength);
+ EXPECT_EQ(0u, receiver_.payload_size());
+ EXPECT_EQ(payload_length, receiver_.rtp_header().header.paddingLength);
}
timestamp += 3000;
fake_clock_.AdvanceTimeMilliseconds(33);
