| /* |
| * 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. |
| */ |
| |
| /* |
| * This file includes unit tests for the RTPSender. |
| */ |
| |
| #include "testing/gtest/include/gtest/gtest.h" |
| |
| #include "webrtc/modules/pacing/include/mock/mock_paced_sender.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/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/system_wrappers/interface/scoped_ptr.h" |
| #include "webrtc/test/mock_transport.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| const int kTransmissionTimeOffsetExtensionId = 1; |
| const int kAbsoluteSendTimeExtensionId = 14; |
| const int kPayload = 100; |
| const uint32_t kTimestamp = 10; |
| const uint16_t kSeqNum = 33; |
| const int kTimeOffset = 22222; |
| const int kMaxPacketLength = 1500; |
| const uint32_t kAbsoluteSendTime = 0x00aabbcc; |
| const uint8_t kAudioLevel = 0x5a; |
| const uint8_t kAudioLevelExtensionId = 9; |
| const int kAudioPayload = 103; |
| const uint64_t kStartTime = 123456789; |
| const size_t kMaxPaddingSize = 224u; |
| } // namespace |
| |
| using testing::_; |
| |
| const uint8_t* GetPayloadData(const RTPHeader& rtp_header, |
| const uint8_t* packet) { |
| return packet + rtp_header.headerLength; |
| } |
| |
| uint16_t GetPayloadDataLength(const RTPHeader& rtp_header, |
| const uint16_t packet_length) { |
| uint16_t length = packet_length - rtp_header.headerLength - |
| rtp_header.paddingLength; |
| return static_cast<uint16_t>(length); |
| } |
| |
| uint64_t ConvertMsToAbsSendTime(int64_t time_ms) { |
| return 0x00fffffful & ((time_ms << 18) / 1000); |
| } |
| |
| class LoopbackTransportTest : public webrtc::Transport { |
| public: |
| LoopbackTransportTest() |
| : packets_sent_(0), last_sent_packet_len_(0), total_bytes_sent_(0) {} |
| virtual int SendPacket(int channel, const void *data, int len) { |
| packets_sent_++; |
| memcpy(last_sent_packet_, data, len); |
| last_sent_packet_len_ = len; |
| total_bytes_sent_ += static_cast<size_t>(len); |
| return len; |
| } |
| virtual int SendRTCPPacket(int channel, const void *data, int len) { |
| return -1; |
| } |
| int packets_sent_; |
| int last_sent_packet_len_; |
| size_t total_bytes_sent_; |
| uint8_t last_sent_packet_[kMaxPacketLength]; |
| }; |
| |
| class RtpSenderTest : public ::testing::Test { |
| protected: |
| RtpSenderTest() |
| : fake_clock_(kStartTime), |
| mock_paced_sender_(), |
| rtp_sender_(), |
| payload_(kPayload), |
| transport_(), |
| kMarkerBit(true) { |
| EXPECT_CALL(mock_paced_sender_, |
| SendPacket(_, _, _, _, _, _)).WillRepeatedly(testing::Return(true)); |
| } |
| |
| virtual void SetUp() { |
| rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL, |
| &mock_paced_sender_, NULL, NULL, NULL)); |
| rtp_sender_->SetSequenceNumber(kSeqNum); |
| } |
| |
| SimulatedClock fake_clock_; |
| MockPacedSender mock_paced_sender_; |
| scoped_ptr<RTPSender> rtp_sender_; |
| int payload_; |
| LoopbackTransportTest transport_; |
| const bool kMarkerBit; |
| uint8_t packet_[kMaxPacketLength]; |
| |
| void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) { |
| EXPECT_EQ(kMarkerBit, rtp_header.markerBit); |
| EXPECT_EQ(payload_, rtp_header.payloadType); |
| EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber); |
| EXPECT_EQ(kTimestamp, rtp_header.timestamp); |
| EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.ssrc); |
| EXPECT_EQ(0, rtp_header.numCSRCs); |
| EXPECT_EQ(0, rtp_header.paddingLength); |
| } |
| |
| 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); |
| |
| // Packet should be stored in a send bucket. |
| EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, |
| payload_length, |
| rtp_length, |
| capture_time_ms, |
| kAllowRetransmission, |
| PacedSender::kNormalPriority)); |
| } |
| }; |
| |
| TEST_F(RtpSenderTest, RegisterRtpTransmissionTimeOffsetHeaderExtension) { |
| EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); |
| EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength, |
| rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset)); |
| EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| } |
| |
| TEST_F(RtpSenderTest, RegisterRtpAbsoluteSendTimeHeaderExtension) { |
| EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); |
| EXPECT_EQ(kRtpOneByteHeaderLength + kAbsoluteSendTimeLength, |
| rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime)); |
| EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| } |
| |
| TEST_F(RtpSenderTest, RegisterRtpAudioLevelHeaderExtension) { |
| EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAudioLevel, kAudioLevelExtensionId)); |
| EXPECT_EQ(kRtpOneByteHeaderLength + kAudioLevelLength, |
| rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( |
| kRtpExtensionAudioLevel)); |
| EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| } |
| |
| TEST_F(RtpSenderTest, RegisterRtpHeaderExtensions) { |
| EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); |
| EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength, |
| rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); |
| EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength + |
| kAbsoluteSendTimeLength, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAudioLevel, kAudioLevelExtensionId)); |
| EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength + |
| kAbsoluteSendTimeLength + kAudioLevelLength, |
| rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset)); |
| EXPECT_EQ(kRtpOneByteHeaderLength + kAbsoluteSendTimeLength + |
| kAudioLevelLength, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime)); |
| EXPECT_EQ(kRtpOneByteHeaderLength + kAudioLevelLength, |
| rtp_sender_->RtpHeaderExtensionTotalLength()); |
| EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension( |
| kRtpExtensionAudioLevel)); |
| EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength()); |
| } |
| |
| TEST_F(RtpSenderTest, BuildRTPPacket) { |
| int32_t length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| kTimestamp, |
| 0); |
| EXPECT_EQ(kRtpHeaderSize, length); |
| |
| // Verify |
| webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); |
| webrtc::RTPHeader rtp_header; |
| |
| const bool valid_rtp_header = rtp_parser.Parse(rtp_header, NULL); |
| |
| ASSERT_TRUE(valid_rtp_header); |
| ASSERT_FALSE(rtp_parser.RTCP()); |
| VerifyRTPHeaderCommon(rtp_header); |
| EXPECT_EQ(length, rtp_header.headerLength); |
| EXPECT_FALSE(rtp_header.extension.hasTransmissionTimeOffset); |
| EXPECT_FALSE(rtp_header.extension.hasAbsoluteSendTime); |
| EXPECT_FALSE(rtp_header.extension.hasAudioLevel); |
| EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset); |
| EXPECT_EQ(0u, rtp_header.extension.absoluteSendTime); |
| EXPECT_EQ(0u, rtp_header.extension.audioLevel); |
| } |
| |
| TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) { |
| EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset)); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); |
| |
| int32_t length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| kTimestamp, |
| 0); |
| EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), |
| length); |
| |
| // Verify |
| webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); |
| webrtc::RTPHeader rtp_header; |
| |
| RtpHeaderExtensionMap map; |
| map.Register(kRtpExtensionTransmissionTimeOffset, |
| kTransmissionTimeOffsetExtensionId); |
| const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); |
| |
| ASSERT_TRUE(valid_rtp_header); |
| ASSERT_FALSE(rtp_parser.RTCP()); |
| VerifyRTPHeaderCommon(rtp_header); |
| EXPECT_EQ(length, rtp_header.headerLength); |
| EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset); |
| EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset); |
| |
| // Parse without map extension |
| webrtc::RTPHeader rtp_header2; |
| const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL); |
| |
| ASSERT_TRUE(valid_rtp_header2); |
| VerifyRTPHeaderCommon(rtp_header2); |
| EXPECT_EQ(length, rtp_header2.headerLength); |
| EXPECT_FALSE(rtp_header2.extension.hasTransmissionTimeOffset); |
| EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset); |
| } |
| |
| TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) { |
| const int kNegTimeOffset = -500; |
| EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kNegTimeOffset)); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); |
| |
| int32_t length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| kTimestamp, |
| 0); |
| EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), |
| length); |
| |
| // Verify |
| webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); |
| webrtc::RTPHeader rtp_header; |
| |
| RtpHeaderExtensionMap map; |
| map.Register(kRtpExtensionTransmissionTimeOffset, |
| kTransmissionTimeOffsetExtensionId); |
| const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); |
| |
| ASSERT_TRUE(valid_rtp_header); |
| ASSERT_FALSE(rtp_parser.RTCP()); |
| VerifyRTPHeaderCommon(rtp_header); |
| EXPECT_EQ(length, rtp_header.headerLength); |
| EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset); |
| EXPECT_EQ(kNegTimeOffset, rtp_header.extension.transmissionTimeOffset); |
| } |
| |
| TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) { |
| EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime)); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); |
| |
| int32_t length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| kTimestamp, |
| 0); |
| EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), |
| length); |
| |
| // Verify |
| webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); |
| webrtc::RTPHeader rtp_header; |
| |
| RtpHeaderExtensionMap map; |
| map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId); |
| const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); |
| |
| ASSERT_TRUE(valid_rtp_header); |
| ASSERT_FALSE(rtp_parser.RTCP()); |
| VerifyRTPHeaderCommon(rtp_header); |
| EXPECT_EQ(length, rtp_header.headerLength); |
| EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime); |
| EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime); |
| |
| // Parse without map extension |
| webrtc::RTPHeader rtp_header2; |
| const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL); |
| |
| ASSERT_TRUE(valid_rtp_header2); |
| VerifyRTPHeaderCommon(rtp_header2); |
| EXPECT_EQ(length, rtp_header2.headerLength); |
| EXPECT_FALSE(rtp_header2.extension.hasAbsoluteSendTime); |
| EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime); |
| } |
| |
| TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) { |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAudioLevel, kAudioLevelExtensionId)); |
| |
| int32_t length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| kTimestamp, |
| 0); |
| EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), |
| length); |
| |
| // Verify |
| webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); |
| webrtc::RTPHeader rtp_header; |
| |
| // Updating audio level is done in RTPSenderAudio, so simulate it here. |
| 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); |
| |
| ASSERT_TRUE(valid_rtp_header); |
| ASSERT_FALSE(rtp_parser.RTCP()); |
| VerifyRTPHeaderCommon(rtp_header); |
| EXPECT_EQ(length, rtp_header.headerLength); |
| EXPECT_TRUE(rtp_header.extension.hasAudioLevel); |
| // Expect kAudioLevel + 0x80 because we set "voiced" to true in the call to |
| // UpdateAudioLevel(), above. |
| EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel); |
| |
| // Parse without map extension |
| webrtc::RTPHeader rtp_header2; |
| const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL); |
| |
| ASSERT_TRUE(valid_rtp_header2); |
| VerifyRTPHeaderCommon(rtp_header2); |
| EXPECT_EQ(length, rtp_header2.headerLength); |
| EXPECT_FALSE(rtp_header2.extension.hasAudioLevel); |
| EXPECT_EQ(0u, rtp_header2.extension.audioLevel); |
| } |
| |
| TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) { |
| EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset)); |
| EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime)); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAudioLevel, kAudioLevelExtensionId)); |
| |
| int32_t length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| kTimestamp, |
| 0); |
| EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(), |
| length); |
| |
| // Verify |
| webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); |
| webrtc::RTPHeader rtp_header; |
| |
| // Updating audio level is done in RTPSenderAudio, so simulate it here. |
| rtp_parser.Parse(rtp_header); |
| rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel); |
| |
| RtpHeaderExtensionMap map; |
| map.Register(kRtpExtensionTransmissionTimeOffset, |
| kTransmissionTimeOffsetExtensionId); |
| map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId); |
| map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId); |
| const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); |
| |
| ASSERT_TRUE(valid_rtp_header); |
| ASSERT_FALSE(rtp_parser.RTCP()); |
| VerifyRTPHeaderCommon(rtp_header); |
| EXPECT_EQ(length, rtp_header.headerLength); |
| EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset); |
| EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime); |
| EXPECT_TRUE(rtp_header.extension.hasAudioLevel); |
| EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset); |
| EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime); |
| EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel); |
| |
| // Parse without map extension |
| webrtc::RTPHeader rtp_header2; |
| const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL); |
| |
| ASSERT_TRUE(valid_rtp_header2); |
| VerifyRTPHeaderCommon(rtp_header2); |
| EXPECT_EQ(length, rtp_header2.headerLength); |
| EXPECT_FALSE(rtp_header2.extension.hasTransmissionTimeOffset); |
| EXPECT_FALSE(rtp_header2.extension.hasAbsoluteSendTime); |
| EXPECT_FALSE(rtp_header2.extension.hasAudioLevel); |
| EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset); |
| EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime); |
| EXPECT_EQ(0u, rtp_header2.extension.audioLevel); |
| } |
| |
| TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) { |
| EXPECT_CALL(mock_paced_sender_, |
| SendPacket(PacedSender::kNormalPriority, _, kSeqNum, _, _, _)). |
| WillOnce(testing::Return(false)); |
| |
| rtp_sender_->SetStorePacketsStatus(true, 10); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); |
| rtp_sender_->SetTargetBitrate(300000); |
| int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); |
| int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| kTimestamp, |
| capture_time_ms); |
| |
| // Packet should be stored in a send bucket. |
| EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, |
| 0, |
| rtp_length, |
| capture_time_ms, |
| kAllowRetransmission, |
| PacedSender::kNormalPriority)); |
| |
| EXPECT_EQ(0, transport_.packets_sent_); |
| |
| const int kStoredTimeInMs = 100; |
| fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs); |
| |
| rtp_sender_->TimeToSendPacket(kSeqNum, capture_time_ms, false); |
| |
| // Process send bucket. Packet should now be sent. |
| EXPECT_EQ(1, transport_.packets_sent_); |
| EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_); |
| // Parse sent packet. |
| webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, |
| rtp_length); |
| webrtc::RTPHeader rtp_header; |
| RtpHeaderExtensionMap map; |
| map.Register(kRtpExtensionTransmissionTimeOffset, |
| kTransmissionTimeOffsetExtensionId); |
| map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId); |
| const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); |
| ASSERT_TRUE(valid_rtp_header); |
| |
| // Verify transmission time offset. |
| EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset); |
| uint64_t expected_send_time = |
| ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds()); |
| EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime); |
| } |
| |
| TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) { |
| EXPECT_CALL(mock_paced_sender_, |
| SendPacket(PacedSender::kNormalPriority, _, kSeqNum, _, _, _)). |
| WillOnce(testing::Return(false)); |
| |
| rtp_sender_->SetStorePacketsStatus(true, 10); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId)); |
| rtp_sender_->SetTargetBitrate(300000); |
| int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); |
| int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| kTimestamp, |
| capture_time_ms); |
| |
| // Packet should be stored in a send bucket. |
| EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, |
| 0, |
| rtp_length, |
| capture_time_ms, |
| kAllowRetransmission, |
| PacedSender::kNormalPriority)); |
| |
| EXPECT_EQ(0, transport_.packets_sent_); |
| |
| EXPECT_CALL(mock_paced_sender_, |
| SendPacket(PacedSender::kHighPriority, _, kSeqNum, _, _, _)). |
| WillOnce(testing::Return(false)); |
| |
| const int kStoredTimeInMs = 100; |
| fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs); |
| |
| EXPECT_EQ(rtp_length, rtp_sender_->ReSendPacket(kSeqNum)); |
| EXPECT_EQ(0, transport_.packets_sent_); |
| |
| rtp_sender_->TimeToSendPacket(kSeqNum, capture_time_ms, false); |
| |
| // Process send bucket. Packet should now be sent. |
| EXPECT_EQ(1, transport_.packets_sent_); |
| EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_); |
| |
| // Parse sent packet. |
| webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, |
| rtp_length); |
| webrtc::RTPHeader rtp_header; |
| RtpHeaderExtensionMap map; |
| map.Register(kRtpExtensionTransmissionTimeOffset, |
| kTransmissionTimeOffsetExtensionId); |
| map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId); |
| const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map); |
| ASSERT_TRUE(valid_rtp_header); |
| |
| // Verify transmission time offset. |
| EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset); |
| uint64_t expected_send_time = |
| ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds()); |
| EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime); |
| } |
| |
| // This test sends 1 regular video packet, then 4 padding packets, and then |
| // 1 more regular packet. |
| TEST_F(RtpSenderTest, SendPadding) { |
| // Make all (non-padding) packets go to send queue. |
| EXPECT_CALL(mock_paced_sender_, |
| SendPacket(PacedSender::kNormalPriority, _, _, _, _, _)). |
| WillRepeatedly(testing::Return(false)); |
| |
| uint16_t seq_num = kSeqNum; |
| uint32_t timestamp = kTimestamp; |
| rtp_sender_->SetStorePacketsStatus(true, 10); |
| int32_t rtp_header_len = kRtpHeaderSize; |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId)); |
| rtp_header_len += 4; // 4 bytes extension. |
| 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. |
| |
| // Create and set up parser. |
| scoped_ptr<webrtc::RtpHeaderParser> rtp_parser( |
| webrtc::RtpHeaderParser::Create()); |
| ASSERT_TRUE(rtp_parser.get() != NULL); |
| rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, |
| kTransmissionTimeOffsetExtensionId); |
| rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, |
| kAbsoluteSendTimeExtensionId); |
| webrtc::RTPHeader rtp_header; |
| |
| rtp_sender_->SetTargetBitrate(300000); |
| int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); |
| int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| timestamp, |
| capture_time_ms); |
| |
| // Packet should be stored in a send bucket. |
| EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, |
| 0, |
| rtp_length, |
| capture_time_ms, |
| kAllowRetransmission, |
| PacedSender::kNormalPriority)); |
| |
| int total_packets_sent = 0; |
| EXPECT_EQ(total_packets_sent, transport_.packets_sent_); |
| |
| const int kStoredTimeInMs = 100; |
| fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs); |
| rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false); |
| // Packet should now be sent. This test doesn't verify the regular video |
| // packet, since it is tested in another test. |
| EXPECT_EQ(++total_packets_sent, transport_.packets_sent_); |
| timestamp += 90 * kStoredTimeInMs; |
| |
| // Send padding 4 times, waiting 50 ms between each. |
| for (int i = 0; i < 4; ++i) { |
| const int kPaddingPeriodMs = 50; |
| const int kPaddingBytes = 100; |
| const int kMaxPaddingLength = 224; // Value taken from rtp_sender.cc. |
| // Padding will be forced to full packets. |
| EXPECT_EQ(kMaxPaddingLength, rtp_sender_->TimeToSendPadding(kPaddingBytes)); |
| |
| // Process send bucket. Padding should now be sent. |
| EXPECT_EQ(++total_packets_sent, transport_.packets_sent_); |
| EXPECT_EQ(kMaxPaddingLength + rtp_header_len, |
| transport_.last_sent_packet_len_); |
| // Parse sent packet. |
| ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, kPaddingBytes, |
| &rtp_header)); |
| |
| // Verify sequence number and timestamp. |
| EXPECT_EQ(seq_num++, rtp_header.sequenceNumber); |
| EXPECT_EQ(timestamp, rtp_header.timestamp); |
| // Verify transmission time offset. |
| EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset); |
| uint64_t expected_send_time = |
| ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds()); |
| EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime); |
| fake_clock_.AdvanceTimeMilliseconds(kPaddingPeriodMs); |
| timestamp += 90 * kPaddingPeriodMs; |
| } |
| |
| // Send a regular video packet again. |
| capture_time_ms = fake_clock_.TimeInMilliseconds(); |
| rtp_length = rtp_sender_->BuildRTPheader(packet_, |
| kPayload, |
| kMarkerBit, |
| timestamp, |
| capture_time_ms); |
| |
| // Packet should be stored in a send bucket. |
| EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, |
| 0, |
| rtp_length, |
| capture_time_ms, |
| kAllowRetransmission, |
| PacedSender::kNormalPriority)); |
| |
| rtp_sender_->TimeToSendPacket(seq_num, capture_time_ms, false); |
| // Process send bucket. |
| 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)); |
| |
| // Verify sequence number and timestamp. |
| EXPECT_EQ(seq_num, rtp_header.sequenceNumber); |
| EXPECT_EQ(timestamp, rtp_header.timestamp); |
| // Verify transmission time offset. This packet is sent without delay. |
| EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset); |
| uint64_t expected_send_time = |
| ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds()); |
| EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime); |
| } |
| |
| TEST_F(RtpSenderTest, SendRedundantPayloads) { |
| MockTransport transport; |
| rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, NULL, |
| &mock_paced_sender_, NULL, NULL, NULL)); |
| rtp_sender_->SetSequenceNumber(kSeqNum); |
| // Make all packets go through the pacer. |
| EXPECT_CALL(mock_paced_sender_, |
| SendPacket(PacedSender::kNormalPriority, _, _, _, _, _)). |
| WillRepeatedly(testing::Return(false)); |
| |
| uint16_t seq_num = kSeqNum; |
| rtp_sender_->SetStorePacketsStatus(true, 10); |
| int32_t rtp_header_len = kRtpHeaderSize; |
| 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. |
| |
| rtp_sender_->SetRTXStatus(kRtxRetransmitted | kRtxRedundantPayloads); |
| rtp_sender_->SetRtxSsrc(1234); |
| |
| // Create and set up parser. |
| scoped_ptr<webrtc::RtpHeaderParser> rtp_parser( |
| webrtc::RtpHeaderParser::Create()); |
| ASSERT_TRUE(rtp_parser.get() != NULL); |
| rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, |
| kTransmissionTimeOffsetExtensionId); |
| rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, |
| 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}; |
| // Send 10 packets of increasing size. |
| for (size_t i = 0; i < kNumPayloadSizes; ++i) { |
| int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); |
| EXPECT_CALL(transport, SendPacket(_, _, _)) |
| .WillOnce(testing::ReturnArg<2>()); |
| SendPacket(capture_time_ms, kPayloadSizes[i]); |
| rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false); |
| fake_clock_.AdvanceTimeMilliseconds(33); |
| } |
| // The amount of padding to send it too small to send a payload packet. |
| EXPECT_CALL(transport, |
| SendPacket(_, _, kMaxPaddingSize + rtp_header_len)) |
| .WillOnce(testing::ReturnArg<2>()); |
| EXPECT_EQ(kMaxPaddingSize, |
| static_cast<size_t>(rtp_sender_->TimeToSendPadding(49))); |
| |
| const int kRtxHeaderSize = 2; |
| EXPECT_CALL(transport, SendPacket(_, _, kPayloadSizes[0] + |
| rtp_header_len + kRtxHeaderSize)) |
| .WillOnce(testing::ReturnArg<2>()); |
| EXPECT_EQ(kPayloadSizes[0], |
| static_cast<size_t>(rtp_sender_->TimeToSendPadding(500))); |
| |
| EXPECT_CALL(transport, SendPacket(_, _, kPayloadSizes[kNumPayloadSizes - 1] + |
| rtp_header_len + kRtxHeaderSize)) |
| .WillOnce(testing::ReturnArg<2>()); |
| EXPECT_CALL(transport, SendPacket(_, _, kMaxPaddingSize + rtp_header_len)) |
| .WillOnce(testing::ReturnArg<2>()); |
| EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize, |
| static_cast<size_t>(rtp_sender_->TimeToSendPadding(999))); |
| } |
| |
| TEST_F(RtpSenderTest, SendGenericVideo) { |
| 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)); |
| uint8_t payload[] = {47, 11, 32, 93, 89}; |
| |
| // Send keyframe |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, |
| 4321, payload, sizeof(payload), |
| NULL)); |
| |
| RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, |
| transport_.last_sent_packet_len_); |
| webrtc::RTPHeader rtp_header; |
| ASSERT_TRUE(rtp_parser.Parse(rtp_header)); |
| |
| 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), |
| GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_)); |
| |
| EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit); |
| EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit); |
| |
| EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload))); |
| |
| // Send delta frame |
| payload[0] = 13; |
| payload[1] = 42; |
| payload[4] = 13; |
| |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type, |
| 1234, 4321, payload, |
| sizeof(payload), NULL)); |
| |
| RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_, |
| transport_.last_sent_packet_len_); |
| ASSERT_TRUE(rtp_parser.Parse(rtp_header)); |
| |
| payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_); |
| generic_header = *payload_data++; |
| |
| EXPECT_FALSE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit); |
| EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit); |
| |
| ASSERT_EQ(sizeof(payload) + sizeof(generic_header), |
| GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_)); |
| |
| EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload))); |
| } |
| |
| TEST_F(RtpSenderTest, FrameCountCallbacks) { |
| class TestCallback : public FrameCountObserver { |
| public: |
| TestCallback() |
| : FrameCountObserver(), num_calls_(0), ssrc_(0), |
| key_frames_(0), delta_frames_(0) {} |
| virtual ~TestCallback() {} |
| |
| virtual void FrameCountUpdated(FrameType frame_type, |
| uint32_t frame_count, |
| const unsigned int ssrc) { |
| ++num_calls_; |
| ssrc_ = ssrc; |
| switch (frame_type) { |
| case kVideoFrameDelta: |
| delta_frames_ = frame_count; |
| break; |
| case kVideoFrameKey: |
| key_frames_ = frame_count; |
| break; |
| default: |
| break; |
| } |
| } |
| |
| uint32_t num_calls_; |
| uint32_t ssrc_; |
| uint32_t key_frames_; |
| uint32_t delta_frames_; |
| } callback; |
| |
| rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL, |
| &mock_paced_sender_, NULL, &callback, NULL)); |
| |
| 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)); |
| uint8_t payload[] = {47, 11, 32, 93, 89}; |
| rtp_sender_->SetStorePacketsStatus(true, 1); |
| uint32_t ssrc = rtp_sender_->SSRC(); |
| |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, |
| 4321, payload, sizeof(payload), |
| NULL)); |
| |
| EXPECT_EQ(1U, callback.num_calls_); |
| EXPECT_EQ(ssrc, callback.ssrc_); |
| EXPECT_EQ(1U, callback.key_frames_); |
| EXPECT_EQ(0U, callback.delta_frames_); |
| |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, |
| payload_type, 1234, 4321, payload, |
| sizeof(payload), NULL)); |
| |
| EXPECT_EQ(2U, callback.num_calls_); |
| EXPECT_EQ(ssrc, callback.ssrc_); |
| EXPECT_EQ(1U, callback.key_frames_); |
| EXPECT_EQ(1U, callback.delta_frames_); |
| |
| rtp_sender_.reset(); |
| } |
| |
| TEST_F(RtpSenderTest, BitrateCallbacks) { |
| class TestCallback : public BitrateStatisticsObserver { |
| public: |
| TestCallback() |
| : BitrateStatisticsObserver(), num_calls_(0), ssrc_(0), bitrate_() {} |
| virtual ~TestCallback() {} |
| |
| virtual void Notify(const BitrateStatistics& stats, uint32_t ssrc) { |
| ++num_calls_; |
| ssrc_ = ssrc; |
| bitrate_ = stats; |
| } |
| |
| uint32_t num_calls_; |
| uint32_t ssrc_; |
| BitrateStatistics bitrate_; |
| } callback; |
| rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL, |
| &mock_paced_sender_, &callback, NULL, NULL)); |
| |
| // Simulate kNumPackets sent with kPacketInterval ms intervals. |
| const uint32_t kNumPackets = 15; |
| const uint32_t kPacketInterval = 20; |
| // Overhead = 12 bytes RTP header + 1 byte generic header. |
| const uint32_t kPacketOverhead = 13; |
| |
| 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)); |
| uint8_t payload[] = {47, 11, 32, 93, 89}; |
| rtp_sender_->SetStorePacketsStatus(true, 1); |
| uint32_t ssrc = rtp_sender_->SSRC(); |
| |
| // Initial process call so we get a new time window. |
| rtp_sender_->ProcessBitrate(); |
| uint64_t start_time = fake_clock_.CurrentNtpInMilliseconds(); |
| |
| // 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)); |
| fake_clock_.AdvanceTimeMilliseconds(kPacketInterval); |
| } |
| |
| rtp_sender_->ProcessBitrate(); |
| |
| const uint32_t expected_packet_rate = 1000 / kPacketInterval; |
| |
| EXPECT_EQ(1U, callback.num_calls_); |
| EXPECT_EQ(ssrc, callback.ssrc_); |
| EXPECT_EQ(start_time + (kNumPackets * kPacketInterval), |
| callback.bitrate_.timestamp_ms); |
| EXPECT_EQ(expected_packet_rate, callback.bitrate_.packet_rate); |
| EXPECT_EQ((kPacketOverhead + sizeof(payload)) * 8 * expected_packet_rate, |
| callback.bitrate_.bitrate_bps); |
| |
| rtp_sender_.reset(); |
| } |
| |
| class RtpSenderAudioTest : public RtpSenderTest { |
| protected: |
| RtpSenderAudioTest() {} |
| |
| virtual void SetUp() { |
| payload_ = kAudioPayload; |
| rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, NULL, |
| &mock_paced_sender_, NULL, NULL, NULL)); |
| rtp_sender_->SetSequenceNumber(kSeqNum); |
| } |
| }; |
| |
| TEST_F(RtpSenderTest, StreamDataCountersCallbacks) { |
| class TestCallback : public StreamDataCountersCallback { |
| public: |
| TestCallback() |
| : StreamDataCountersCallback(), ssrc_(0), counters_() {} |
| virtual ~TestCallback() {} |
| |
| virtual void DataCountersUpdated(const StreamDataCounters& counters, |
| uint32_t ssrc) { |
| ssrc_ = ssrc; |
| counters_ = counters; |
| } |
| |
| uint32_t ssrc_; |
| StreamDataCounters counters_; |
| bool Matches(uint32_t ssrc, uint32_t bytes, uint32_t header_bytes, |
| uint32_t padding, uint32_t packets, uint32_t retransmits, |
| uint32_t fec) { |
| return ssrc_ == ssrc && |
| counters_.bytes == bytes && |
| counters_.header_bytes == header_bytes && |
| counters_.padding_bytes == padding && |
| counters_.packets == packets && |
| counters_.retransmitted_packets == retransmits && |
| counters_.fec_packets == fec; |
| } |
| |
| } callback; |
| |
| const uint8_t kRedPayloadType = 96; |
| const uint8_t kUlpfecPayloadType = 97; |
| 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)); |
| uint8_t payload[] = {47, 11, 32, 93, 89}; |
| rtp_sender_->SetStorePacketsStatus(true, 1); |
| uint32_t ssrc = rtp_sender_->SSRC(); |
| |
| rtp_sender_->RegisterRtpStatisticsCallback(&callback); |
| |
| // Send a frame. |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, |
| 4321, payload, sizeof(payload), |
| NULL)); |
| |
| // {bytes = 6, header = 12, padding = 0, packets = 1, retrans = 0, fec = 0} |
| EXPECT_TRUE(callback.Matches(ssrc, 6, 12, 0, 1, 0, 0)); |
| |
| // Retransmit a frame. |
| uint16_t seqno = rtp_sender_->SequenceNumber() - 1; |
| rtp_sender_->ReSendPacket(seqno, 0); |
| |
| // bytes = 6, header = 12, padding = 0, packets = 2, retrans = 1, fec = 0} |
| EXPECT_TRUE(callback.Matches(ssrc, 6, 12, 0, 2, 1, 0)); |
| |
| // Send padding. |
| rtp_sender_->TimeToSendPadding(kMaxPaddingSize); |
| // {bytes = 6, header = 24, padding = 224, packets = 3, retrans = 1, fec = 0} |
| EXPECT_TRUE(callback.Matches(ssrc, 6, 24, kMaxPaddingSize, 3, 1, 0)); |
| |
| // Send FEC. |
| rtp_sender_->SetGenericFECStatus(true, kRedPayloadType, kUlpfecPayloadType); |
| FecProtectionParams fec_params; |
| fec_params.fec_mask_type = kFecMaskRandom; |
| fec_params.fec_rate = 1; |
| fec_params.max_fec_frames = 1; |
| fec_params.use_uep_protection = false; |
| rtp_sender_->SetFecParameters(&fec_params, &fec_params); |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type, |
| 1234, 4321, payload, |
| sizeof(payload), NULL)); |
| |
| // {bytes = 34, header = 48, padding = 224, packets = 5, retrans = 1, fec = 1} |
| EXPECT_TRUE(callback.Matches(ssrc, 34, 48, kMaxPaddingSize, 5, 1, 1)); |
| |
| rtp_sender_->RegisterRtpStatisticsCallback(NULL); |
| } |
| |
| TEST_F(RtpSenderAudioTest, SendAudio) { |
| char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; |
| const uint8_t payload_type = 127; |
| ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000, |
| 0, 1500)); |
| uint8_t payload[] = {47, 11, 32, 93, 89}; |
| |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, |
| 4321, payload, sizeof(payload), |
| NULL)); |
| |
| RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, |
| transport_.last_sent_packet_len_); |
| webrtc::RTPHeader rtp_header; |
| ASSERT_TRUE(rtp_parser.Parse(rtp_header)); |
| |
| 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))); |
| } |
| |
| TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) { |
| EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel)); |
| EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
| kRtpExtensionAudioLevel, kAudioLevelExtensionId)); |
| |
| char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; |
| const uint8_t payload_type = 127; |
| ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000, |
| 0, 1500)); |
| uint8_t payload[] = {47, 11, 32, 93, 89}; |
| |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, |
| 4321, payload, sizeof(payload), |
| NULL)); |
| |
| RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, |
| transport_.last_sent_packet_len_); |
| webrtc::RTPHeader rtp_header; |
| ASSERT_TRUE(rtp_parser.Parse(rtp_header)); |
| |
| 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. |
| }; |
| |
| EXPECT_EQ(0, memcmp(extension, payload_data - sizeof(extension), |
| sizeof(extension))); |
| } |
| |
| // As RFC4733, named telephone events are carried as part of the audio stream |
| // and must use the same sequence number and timestamp base as the regular |
| // audio channel. |
| // This test checks the marker bit for the first packet and the consequent |
| // packets of the same telephone event. Since it is specifically for DTMF |
| // events, ignoring audio packets and sending kFrameEmpty instead of those. |
| 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)); |
| // 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)); |
| int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); |
| // DTMF event key=9, duration=500 and attenuationdB=10 |
| rtp_sender_->SendTelephoneEvent(9, 500, 10); |
| // During start, it takes the starting timestamp as last sent timestamp. |
| // The duration is calculated as the difference of current and last sent |
| // timestamp. So for first call it will skip since the duration is zero. |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type, |
| capture_time_ms, |
| 0, NULL, 0, |
| NULL)); |
| // DTMF Sample Length is (Frequency/1000) * Duration. |
| // So in this case, it is (8000/1000) * 500 = 4000. |
| // Sending it as two packets. |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type, |
| capture_time_ms+2000, |
| 0, NULL, 0, |
| NULL)); |
| scoped_ptr<webrtc::RtpHeaderParser> rtp_parser( |
| webrtc::RtpHeaderParser::Create()); |
| ASSERT_TRUE(rtp_parser.get() != NULL); |
| webrtc::RTPHeader rtp_header; |
| ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, |
| transport_.last_sent_packet_len_, |
| &rtp_header)); |
| // Marker Bit should be set to 1 for first packet. |
| EXPECT_TRUE(rtp_header.markerBit); |
| |
| ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type, |
| capture_time_ms+4000, |
| 0, NULL, 0, |
| NULL)); |
| ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, |
| transport_.last_sent_packet_len_, |
| &rtp_header)); |
| // Marker Bit should be set to 0 for rest of the packets. |
| EXPECT_FALSE(rtp_header.markerBit); |
| } |
| |
| TEST_F(RtpSenderTest, BytesReportedCorrectly) { |
| const char* kPayloadName = "GENERIC"; |
| const uint8_t kPayloadType = 127; |
| rtp_sender_->SetSSRC(1234); |
| rtp_sender_->SetRtxSsrc(4321); |
| rtp_sender_->SetRtxPayloadType(kPayloadType - 1); |
| rtp_sender_->SetRTXStatus(kRtxRetransmitted | kRtxRedundantPayloads); |
| |
| 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)); |
| |
| // Will send 2 full-size padding packets. |
| rtp_sender_->TimeToSendPadding(1); |
| rtp_sender_->TimeToSendPadding(1); |
| |
| StreamDataCounters rtp_stats; |
| StreamDataCounters rtx_stats; |
| rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats); |
| |
| // Payload + 1-byte generic header. |
| EXPECT_EQ(rtp_stats.bytes, sizeof(payload) + 1); |
| EXPECT_EQ(rtp_stats.header_bytes, 12u); |
| EXPECT_EQ(rtp_stats.padding_bytes, 0u); |
| EXPECT_EQ(rtx_stats.bytes, 0u); |
| EXPECT_EQ(rtx_stats.header_bytes, 24u); |
| EXPECT_EQ(rtx_stats.padding_bytes, 2 * kMaxPaddingSize); |
| |
| EXPECT_EQ(transport_.total_bytes_sent_, |
| rtp_stats.bytes + rtp_stats.header_bytes + rtp_stats.padding_bytes + |
| rtx_stats.bytes + rtx_stats.header_bytes + |
| rtx_stats.padding_bytes); |
| } |
| } // namespace webrtc |
