diff options
Diffstat (limited to 'webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc')
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc | 270 |
1 files changed, 134 insertions, 136 deletions
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc index fde6d47ceb..6bc122201a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc @@ -12,20 +12,22 @@ * This file includes unit tests for the RTPSender. */ +#include <list> +#include <vector> + #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" - #include "webrtc/base/buffer.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" -#include "webrtc/system_wrappers/include/stl_util.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" +#include "webrtc/system_wrappers/include/stl_util.h" #include "webrtc/test/mock_transport.h" #include "webrtc/typedefs.h" @@ -90,9 +92,7 @@ class LoopbackTransportTest : public webrtc::Transport { sent_packets_.push_back(buffer); return true; } - bool SendRtcp(const uint8_t* data, size_t len) override { - return false; - } + bool SendRtcp(const uint8_t* data, size_t len) override { return false; } int packets_sent_; size_t last_sent_packet_len_; size_t total_bytes_sent_; @@ -163,11 +163,8 @@ class RtpSenderTest : public ::testing::Test { void SendPacket(int64_t capture_time_ms, int payload_length) { uint32_t timestamp = capture_time_ms * 90; - int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_, - kPayload, - kMarkerBit, - timestamp, - capture_time_ms); + int32_t rtp_length = rtp_sender_->BuildRTPheader( + packet_, kPayload, kMarkerBit, timestamp, capture_time_ms); ASSERT_GE(rtp_length, 0); // Packet should be stored in a send bucket. @@ -186,7 +183,7 @@ class RtpSenderTestWithoutPacer : public RtpSenderTest { class RtpSenderVideoTest : public RtpSenderTest { protected: - virtual void SetUp() override { + void SetUp() override { // TODO(pbos): Set up to use pacer. SetUpRtpSender(false); rtp_sender_video_.reset( @@ -211,7 +208,7 @@ class RtpSenderVideoTest : public RtpSenderTest { } else { ASSERT_EQ(kRtpHeaderSize, length); } - ASSERT_TRUE(rtp_parser.Parse(rtp_header, map)); + ASSERT_TRUE(rtp_parser.Parse(&rtp_header, map)); ASSERT_FALSE(rtp_parser.RTCP()); EXPECT_EQ(payload_, rtp_header.payloadType); EXPECT_EQ(seq_num, rtp_header.sequenceNumber); @@ -228,53 +225,57 @@ TEST_F(RtpSenderTestWithoutPacer, RegisterRtpTransmissionTimeOffsetHeaderExtension) { EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); + kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId)); EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength, rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset)); + kRtpExtensionTransmissionTimeOffset)); EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength()); } TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAbsoluteSendTimeHeaderExtension) { EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength()); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); + EXPECT_EQ( + 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + kAbsoluteSendTimeExtensionId)); EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAbsoluteSendTimeLength), rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime)); + kRtpExtensionAbsoluteSendTime)); EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength()); } TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAudioLevelHeaderExtension) { EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength()); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAudioLevel, kAudioLevelExtensionId)); + EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, + kAudioLevelExtensionId)); EXPECT_EQ( RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAudioLevelLength), rtp_sender_->RtpHeaderExtensionTotalLength()); - EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( - kRtpExtensionAudioLevel)); + EXPECT_EQ(0, + rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel)); EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength()); } TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) { EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); + kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId)); EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength), rtp_sender_->RtpHeaderExtensionTotalLength()); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); + EXPECT_EQ( + 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + kAbsoluteSendTimeExtensionId)); EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength + kAbsoluteSendTimeLength), rtp_sender_->RtpHeaderExtensionTotalLength()); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAudioLevel, kAudioLevelExtensionId)); + EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, + kAudioLevelExtensionId)); EXPECT_EQ(RtpUtility::Word32Align( kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength + kAbsoluteSendTimeLength + kAudioLevelLength), @@ -290,18 +291,18 @@ TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) { // Deregister starts. EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset)); + kRtpExtensionTransmissionTimeOffset)); EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAbsoluteSendTimeLength + kAudioLevelLength + kVideoRotationLength), rtp_sender_->RtpHeaderExtensionTotalLength()); EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime)); + kRtpExtensionAbsoluteSendTime)); EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAudioLevelLength + kVideoRotationLength), rtp_sender_->RtpHeaderExtensionTotalLength()); - EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( - kRtpExtensionAudioLevel)); + EXPECT_EQ(0, + rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel)); EXPECT_EQ( RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength), rtp_sender_->RtpHeaderExtensionTotalLength()); @@ -334,7 +335,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) { webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, nullptr); + const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, nullptr); ASSERT_TRUE(valid_rtp_header); ASSERT_FALSE(rtp_parser.RTCP()); @@ -354,7 +355,8 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithTransmissionOffsetExtension) { EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset)); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); + kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId)); size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader( packet_, kPayload, kMarkerBit, kTimestamp, 0)); @@ -368,7 +370,7 @@ TEST_F(RtpSenderTestWithoutPacer, RtpHeaderExtensionMap map; map.Register(kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId); - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); + const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map); ASSERT_TRUE(valid_rtp_header); ASSERT_FALSE(rtp_parser.RTCP()); @@ -379,7 +381,7 @@ TEST_F(RtpSenderTestWithoutPacer, // Parse without map extension webrtc::RTPHeader rtp_header2; - const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr); + const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr); ASSERT_TRUE(valid_rtp_header2); VerifyRTPHeaderCommon(rtp_header2); @@ -393,7 +395,8 @@ TEST_F(RtpSenderTestWithoutPacer, const int kNegTimeOffset = -500; EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kNegTimeOffset)); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); + kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId)); size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader( packet_, kPayload, kMarkerBit, kTimestamp, 0)); @@ -407,7 +410,7 @@ TEST_F(RtpSenderTestWithoutPacer, RtpHeaderExtensionMap map; map.Register(kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId); - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); + const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map); ASSERT_TRUE(valid_rtp_header); ASSERT_FALSE(rtp_parser.RTCP()); @@ -419,8 +422,9 @@ TEST_F(RtpSenderTestWithoutPacer, TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) { EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime)); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); + EXPECT_EQ( + 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + kAbsoluteSendTimeExtensionId)); size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader( packet_, kPayload, kMarkerBit, kTimestamp, 0)); @@ -433,7 +437,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) { RtpHeaderExtensionMap map; map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId); - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); + const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map); ASSERT_TRUE(valid_rtp_header); ASSERT_FALSE(rtp_parser.RTCP()); @@ -444,7 +448,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) { // Parse without map extension webrtc::RTPHeader rtp_header2; - const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr); + const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr); ASSERT_TRUE(valid_rtp_header2); VerifyRTPHeaderCommon(rtp_header2); @@ -472,7 +476,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) { webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; - ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map)); + ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map)); ASSERT_FALSE(rtp_parser.RTCP()); VerifyRTPHeaderCommon(rtp_header); EXPECT_EQ(length, rtp_header.headerLength); @@ -500,7 +504,7 @@ TEST_F(RtpSenderTestWithoutPacer, webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; - ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map)); + ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map)); ASSERT_FALSE(rtp_parser.RTCP()); VerifyRTPHeaderCommon(rtp_header, false); EXPECT_EQ(length, rtp_header.headerLength); @@ -508,8 +512,8 @@ TEST_F(RtpSenderTestWithoutPacer, } TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) { - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAudioLevel, kAudioLevelExtensionId)); + EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, + kAudioLevelExtensionId)); size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader( packet_, kPayload, kMarkerBit, kTimestamp, 0)); @@ -521,12 +525,12 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) { webrtc::RTPHeader rtp_header; // Updating audio level is done in RTPSenderAudio, so simulate it here. - rtp_parser.Parse(rtp_header); + rtp_parser.Parse(&rtp_header); rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel); RtpHeaderExtensionMap map; map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId); - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); + const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map); ASSERT_TRUE(valid_rtp_header); ASSERT_FALSE(rtp_parser.RTCP()); @@ -538,7 +542,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) { // Parse without map extension webrtc::RTPHeader rtp_header2; - const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr); + const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr); ASSERT_TRUE(valid_rtp_header2); VerifyRTPHeaderCommon(rtp_header2); @@ -554,11 +558,13 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) { EXPECT_EQ(0, rtp_sender_->SetTransportSequenceNumber(kTransportSequenceNumber)); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAudioLevel, kAudioLevelExtensionId)); + kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId)); + EXPECT_EQ( + 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + kAbsoluteSendTimeExtensionId)); + EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, + kAudioLevelExtensionId)); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionTransportSequenceNumber, kTransportSequenceNumberExtensionId)); @@ -573,7 +579,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) { webrtc::RTPHeader rtp_header; // Updating audio level is done in RTPSenderAudio, so simulate it here. - rtp_parser.Parse(rtp_header); + rtp_parser.Parse(&rtp_header); rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel); RtpHeaderExtensionMap map; @@ -583,7 +589,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) { map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId); map.Register(kRtpExtensionTransportSequenceNumber, kTransportSequenceNumberExtensionId); - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); + const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map); ASSERT_TRUE(valid_rtp_header); ASSERT_FALSE(rtp_parser.RTCP()); @@ -602,7 +608,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) { // Parse without map extension webrtc::RTPHeader rtp_header2; - const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr); + const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr); ASSERT_TRUE(valid_rtp_header2); VerifyRTPHeaderCommon(rtp_header2); @@ -626,9 +632,11 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) { rtp_sender_->SetStorePacketsStatus(true, 10); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); + kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId)); + EXPECT_EQ( + 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + kAbsoluteSendTimeExtensionId)); rtp_sender_->SetTargetBitrate(300000); int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); int rtp_length_int = rtp_sender_->BuildRTPheader( @@ -659,7 +667,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) { map.Register(kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId); map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId); - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); + const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map); ASSERT_TRUE(valid_rtp_header); // Verify transmission time offset. @@ -676,9 +684,11 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) { rtp_sender_->SetStorePacketsStatus(true, 10); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); + kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId)); + EXPECT_EQ( + 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + kAbsoluteSendTimeExtensionId)); rtp_sender_->SetTargetBitrate(300000); int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); int rtp_length_int = rtp_sender_->BuildRTPheader( @@ -717,7 +727,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) { map.Register(kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId); map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId); - const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); + const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map); ASSERT_TRUE(valid_rtp_header); // Verify transmission time offset. @@ -740,10 +750,12 @@ TEST_F(RtpSenderTest, SendPadding) { rtp_sender_->SetStorePacketsStatus(true, 10); size_t rtp_header_len = kRtpHeaderSize; EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); + kRtpExtensionTransmissionTimeOffset, + kTransmissionTimeOffsetExtensionId)); rtp_header_len += 4; // 4 bytes extension. - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); + EXPECT_EQ( + 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + kAbsoluteSendTimeExtensionId)); rtp_header_len += 4; // 4 bytes extension. rtp_header_len += 4; // 4 extra bytes common to all extension headers. @@ -815,8 +827,8 @@ TEST_F(RtpSenderTest, SendPadding) { // Send a regular video packet again. capture_time_ms = fake_clock_.TimeInMilliseconds(); - rtp_length_int = rtp_sender_->BuildRTPheader( - packet_, kPayload, kMarkerBit, timestamp, capture_time_ms); + rtp_length_int = rtp_sender_->BuildRTPheader(packet_, kPayload, kMarkerBit, + timestamp, capture_time_ms); ASSERT_NE(-1, rtp_length_int); rtp_length = static_cast<size_t>(rtp_length_int); @@ -830,8 +842,8 @@ TEST_F(RtpSenderTest, SendPadding) { EXPECT_EQ(++total_packets_sent, transport_.packets_sent_); EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_); // Parse sent packet. - ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, rtp_length, - &rtp_header)); + ASSERT_TRUE( + rtp_parser->Parse(transport_.last_sent_packet_, rtp_length, &rtp_header)); // Verify sequence number and timestamp. EXPECT_EQ(seq_num, rtp_header.sequenceNumber); @@ -858,8 +870,9 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) { uint16_t seq_num = kSeqNum; rtp_sender_->SetStorePacketsStatus(true, 10); int32_t rtp_header_len = kRtpHeaderSize; - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); + EXPECT_EQ( + 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, + kAbsoluteSendTimeExtensionId)); rtp_header_len += 4; // 4 bytes extension. rtp_header_len += 4; // 4 extra bytes common to all extension headers. @@ -876,8 +889,8 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) { kAbsoluteSendTimeExtensionId); rtp_sender_->SetTargetBitrate(300000); const size_t kNumPayloadSizes = 10; - const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, 750, - 800, 850, 900, 950}; + const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, + 750, 800, 850, 900, 950}; // Send 10 packets of increasing size. for (size_t i = 0; i < kNumPayloadSizes; ++i) { int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); @@ -921,10 +934,10 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) { RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, transport_.last_sent_packet_len_); webrtc::RTPHeader rtp_header; - ASSERT_TRUE(rtp_parser.Parse(rtp_header)); + ASSERT_TRUE(rtp_parser.Parse(&rtp_header)); - const uint8_t* payload_data = GetPayloadData(rtp_header, - transport_.last_sent_packet_); + const uint8_t* payload_data = + GetPayloadData(rtp_header, transport_.last_sent_packet_); uint8_t generic_header = *payload_data++; ASSERT_EQ(sizeof(payload) + sizeof(generic_header), @@ -946,7 +959,7 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) { RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_, transport_.last_sent_packet_len_); - ASSERT_TRUE(rtp_parser.Parse(rtp_header)); + ASSERT_TRUE(rtp_parser.Parse(&rtp_header)); payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_); generic_header = *payload_data++; @@ -1043,9 +1056,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) { char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; const uint8_t payload_type = 127; - ASSERT_EQ( - 0, - rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 0, 1500)); + ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, + 0, 1500)); uint8_t payload[] = {47, 11, 32, 93, 89}; rtp_sender_->SetStorePacketsStatus(true, 1); uint32_t ssrc = rtp_sender_->SSRC(); @@ -1057,13 +1069,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) { // Send a few frames. for (uint32_t i = 0; i < kNumPackets; ++i) { ASSERT_EQ(0, - rtp_sender_->SendOutgoingData(kVideoFrameKey, - payload_type, - 1234, - 4321, - payload, - sizeof(payload), - 0)); + rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, + 4321, payload, sizeof(payload), 0)); fake_clock_.AdvanceTimeMilliseconds(kPacketInterval); } @@ -1100,8 +1107,7 @@ class RtpSenderAudioTest : public RtpSenderTest { TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) { class TestCallback : public StreamDataCountersCallback { public: - TestCallback() - : StreamDataCountersCallback(), ssrc_(0), counters_() {} + TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {} virtual ~TestCallback() {} void DataCountersUpdated(const StreamDataCounters& counters, @@ -1127,7 +1133,6 @@ TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) { MatchPacketCounter(counters.retransmitted, counters_.retransmitted); EXPECT_EQ(counters.fec.packets, counters_.fec.packets); } - } callback; const uint8_t kRedPayloadType = 96; @@ -1212,10 +1217,10 @@ TEST_F(RtpSenderAudioTest, SendAudio) { RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, transport_.last_sent_packet_len_); webrtc::RTPHeader rtp_header; - ASSERT_TRUE(rtp_parser.Parse(rtp_header)); + ASSERT_TRUE(rtp_parser.Parse(&rtp_header)); - const uint8_t* payload_data = GetPayloadData(rtp_header, - transport_.last_sent_packet_); + const uint8_t* payload_data = + GetPayloadData(rtp_header, transport_.last_sent_packet_); ASSERT_EQ(sizeof(payload), GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_)); @@ -1225,8 +1230,8 @@ TEST_F(RtpSenderAudioTest, SendAudio) { TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) { EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel)); - EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( - kRtpExtensionAudioLevel, kAudioLevelExtensionId)); + EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, + kAudioLevelExtensionId)); char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; const uint8_t payload_type = 127; @@ -1241,21 +1246,22 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) { RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, transport_.last_sent_packet_len_); webrtc::RTPHeader rtp_header; - ASSERT_TRUE(rtp_parser.Parse(rtp_header)); + ASSERT_TRUE(rtp_parser.Parse(&rtp_header)); - const uint8_t* payload_data = GetPayloadData(rtp_header, - transport_.last_sent_packet_); + const uint8_t* payload_data = + GetPayloadData(rtp_header, transport_.last_sent_packet_); ASSERT_EQ(sizeof(payload), GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_)); EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload))); - uint8_t extension[] = { 0xbe, 0xde, 0x00, 0x01, - (kAudioLevelExtensionId << 4) + 0, // ID + length. - kAudioLevel, // Data. - 0x00, 0x00 // Padding. - }; + uint8_t extension[] = { + 0xbe, 0xde, 0x00, 0x01, + (kAudioLevelExtensionId << 4) + 0, // ID + length. + kAudioLevel, // Data. + 0x00, 0x00 // Padding. + }; EXPECT_EQ(0, memcmp(extension, payload_data - sizeof(extension), sizeof(extension))); @@ -1270,14 +1276,14 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) { TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) { char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event"; uint8_t payload_type = 126; - ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 0, - 0, 0)); + ASSERT_EQ(0, + rtp_sender_->RegisterPayload(payload_name, payload_type, 0, 0, 0)); // For Telephone events, payload is not added to the registered payload list, // it will register only the payload used for audio stream. // Registering the payload again for audio stream with different payload name. - strcpy(payload_name, "payload_name"); - ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 8000, - 1, 0)); + const char kPayloadName[] = "payload_name"; + ASSERT_EQ( + 0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0)); int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); // DTMF event key=9, duration=500 and attenuationdB=10 rtp_sender_->SendTelephoneEvent(9, 500, 10); @@ -1298,8 +1304,7 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) { ASSERT_TRUE(rtp_parser.get() != nullptr); webrtc::RTPHeader rtp_header; ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, - transport_.last_sent_packet_len_, - &rtp_header)); + transport_.last_sent_packet_len_, &rtp_header)); // Marker Bit should be set to 1 for first packet. EXPECT_TRUE(rtp_header.markerBit); @@ -1307,8 +1312,7 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) { capture_time_ms + 4000, 0, nullptr, 0, nullptr)); ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, - transport_.last_sent_packet_len_, - &rtp_header)); + transport_.last_sent_packet_len_, &rtp_header)); // Marker Bit should be set to 0 for rest of the packets. EXPECT_FALSE(rtp_header.markerBit); } @@ -1321,19 +1325,13 @@ TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) { rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType); rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads); - ASSERT_EQ( - 0, - rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000, 0, 1500)); + ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000, + 0, 1500)); uint8_t payload[] = {47, 11, 32, 93, 89}; - ASSERT_EQ(0, - rtp_sender_->SendOutgoingData(kVideoFrameKey, - kPayloadType, - 1234, - 4321, - payload, - sizeof(payload), - 0)); + ASSERT_EQ( + 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234, 4321, + payload, sizeof(payload), 0)); // Will send 2 full-size padding packets. rtp_sender_->TimeToSendPadding(1); @@ -1353,17 +1351,17 @@ TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) { EXPECT_EQ(rtx_stats.transmitted.padding_bytes, 2 * kMaxPaddingSize); EXPECT_EQ(rtp_stats.transmitted.TotalBytes(), - rtp_stats.transmitted.payload_bytes + - rtp_stats.transmitted.header_bytes + - rtp_stats.transmitted.padding_bytes); + rtp_stats.transmitted.payload_bytes + + rtp_stats.transmitted.header_bytes + + rtp_stats.transmitted.padding_bytes); EXPECT_EQ(rtx_stats.transmitted.TotalBytes(), - rtx_stats.transmitted.payload_bytes + - rtx_stats.transmitted.header_bytes + - rtx_stats.transmitted.padding_bytes); + rtx_stats.transmitted.payload_bytes + + rtx_stats.transmitted.header_bytes + + rtx_stats.transmitted.padding_bytes); - EXPECT_EQ(transport_.total_bytes_sent_, - rtp_stats.transmitted.TotalBytes() + - rtx_stats.transmitted.TotalBytes()); + EXPECT_EQ( + transport_.total_bytes_sent_, + rtp_stats.transmitted.TotalBytes() + rtx_stats.transmitted.TotalBytes()); } TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) { |