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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Issue 2399463009: Delete RTPSender::BuildRtpHeader function (Closed)
Patch Set: . Created 4 years, 2 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 22ad119c6f2717c26d5cd5bfd78a57d77b82d655..62fd743d2fb432c62a7f91e4562472e6d4af55e8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -41,9 +41,7 @@ const int kPayload = 100;
const int kRtxPayload = 98;
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 uint16_t kTransportSequenceNumber = 0xaabbu;
const uint8_t kAudioLevelExtensionId = 9;
@@ -51,7 +49,6 @@ const int kAudioPayload = 103;
const uint64_t kStartTime = 123456789;
const size_t kMaxPaddingSize = 224u;
const int kVideoRotationExtensionId = 5;
-const VideoRotation kRotation = kVideoRotation_270;
const size_t kGenericHeaderLength = 1;
const uint8_t kPayloadData[] = {47, 11, 32, 93, 89};
@@ -176,7 +173,6 @@ class RtpSenderTest : public ::testing::Test {
int payload_;
LoopbackTransportTest transport_;
const bool kMarkerBit;
- uint8_t packet_[kMaxPacketLength];
void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
VerifyRTPHeaderCommon(rtp_header, kMarkerBit, 0);
@@ -198,16 +194,29 @@ class RtpSenderTest : public ::testing::Test {
EXPECT_EQ(0U, rtp_header.paddingLength);
}
+ std::unique_ptr<RtpPacketToSend> BuildRtpPacket(int payload_type,
+ bool marker_bit,
+ uint32_t timestamp,
+ int64_t capture_time_ms) {
+ auto packet = rtp_sender_->AllocatePacket();
+ packet->SetPayloadType(payload_type);
+ packet->SetMarker(marker_bit);
+ packet->SetTimestamp(timestamp);
+ packet->set_capture_time_ms(capture_time_ms);
+ EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
+ return packet;
+ }
+
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);
- ASSERT_GE(rtp_length, 0);
+ auto packet =
+ BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
+ packet->AllocatePayload(payload_length);
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(
- packet_, payload_length, rtp_length, capture_time_ms,
- kAllowRetransmission, RtpPacketSender::kNormalPriority));
+ EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
+ kAllowRetransmission,
+ RtpPacketSender::kNormalPriority));
}
void SendGenericPayload() {
@@ -250,14 +259,6 @@ class RtpSenderVideoTest : public RtpSenderTest {
webrtc::RtpUtility::RtpHeaderParser rtp_parser(data, len);
webrtc::RTPHeader rtp_header;
- size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
- packet_, kPayload, expect_cvo /* marker_bit */, kTimestamp, 0));
- if (expect_cvo) {
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(),
- length);
- } else {
- ASSERT_EQ(kRtpHeaderSize, length);
- }
ASSERT_TRUE(rtp_parser.Parse(&rtp_header, map));
ASSERT_FALSE(rtp_parser.RTCP());
EXPECT_EQ(payload_, rtp_header.payloadType);
@@ -470,134 +471,6 @@ TEST_F(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
EXPECT_EQ(kTimestamp, padding_packet.Timestamp());
}
-TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) {
- size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_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, nullptr);
-
- 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_FALSE(rtp_header.extension.voiceActivity);
- EXPECT_EQ(0u, rtp_header.extension.audioLevel);
- EXPECT_EQ(0u, rtp_header.extension.videoRotation);
-}
-
-TEST_F(RtpSenderTestWithoutPacer,
- BuildRTPPacketWithTransmissionOffsetExtension) {
- EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset,
- kTransmissionTimeOffsetExtensionId));
-
- size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), 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, nullptr);
-
- 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(RtpSenderTestWithoutPacer,
- BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
- const int kNegTimeOffset = -500;
- EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kNegTimeOffset));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset,
- kTransmissionTimeOffsetExtensionId));
-
- size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), 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(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
- EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
- EXPECT_EQ(
- 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
- kAbsoluteSendTimeExtensionId));
-
- size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), 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, nullptr);
-
- 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(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, nullptr,
@@ -652,204 +525,6 @@ TEST_F(RtpSenderTestWithoutPacer, OnSendPacketUpdated) {
SendGenericPayload();
}
-// Test CVO header extension is only set when marker bit is true.
-TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
- rtp_sender_->SetVideoRotation(kRotation);
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionVideoRotation, kVideoRotationExtensionId));
- EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
-
- RtpHeaderExtensionMap map;
- map.Register(kRtpExtensionVideoRotation, kVideoRotationExtensionId);
-
- size_t length = static_cast<size_t>(
- rtp_sender_->BuildRtpHeader(packet_, kPayload, true, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
-
- // Verify
- webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
- webrtc::RTPHeader rtp_header;
-
- ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
- ASSERT_FALSE(rtp_parser.RTCP());
- VerifyRTPHeaderCommon(rtp_header);
- EXPECT_EQ(length, rtp_header.headerLength);
- EXPECT_TRUE(rtp_header.extension.hasVideoRotation);
- EXPECT_EQ(kRotation, rtp_header.extension.videoRotation);
-}
-
-// Test CVO header extension is not set when marker bit is false.
-TEST_F(RtpSenderTestWithoutPacer,
- DISABLED_BuildRTPPacketWithVideoRotation_NoMarkerBit) {
- rtp_sender_->SetVideoRotation(kRotation);
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionVideoRotation, kVideoRotationExtensionId));
- EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
-
- RtpHeaderExtensionMap map;
- map.Register(kRtpExtensionVideoRotation, kVideoRotationExtensionId);
-
- size_t length = static_cast<size_t>(
- rtp_sender_->BuildRtpHeader(packet_, kPayload, false, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize, length);
-
- // Verify
- webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
- webrtc::RTPHeader rtp_header;
-
- ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
- ASSERT_FALSE(rtp_parser.RTCP());
- VerifyRTPHeaderCommon(rtp_header, false);
- EXPECT_EQ(length, rtp_header.headerLength);
- EXPECT_FALSE(rtp_header.extension.hasVideoRotation);
-}
-
-TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
- kAudioLevelExtensionId));
-
- size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), 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_TRUE(rtp_header.extension.voiceActivity);
- EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
-
- // Parse without map extension
- webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
-
- ASSERT_TRUE(valid_rtp_header2);
- VerifyRTPHeaderCommon(rtp_header2);
- EXPECT_EQ(length, rtp_header2.headerLength);
- EXPECT_FALSE(rtp_header2.extension.hasAudioLevel);
- EXPECT_FALSE(rtp_header2.extension.voiceActivity);
- EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
-}
-
-TEST_F(RtpSenderTestWithoutPacer,
- BuildRTPPacketWithCSRCAndAudioLevelExtension) {
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
- kAudioLevelExtensionId));
- std::vector<uint32_t> csrcs;
- csrcs.push_back(0x23456789);
- rtp_sender_->SetCsrcs(csrcs);
- size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, kTimestamp, 0));
-
- // 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);
- EXPECT_TRUE(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, kMarkerBit, csrcs.size());
- EXPECT_EQ(length, rtp_header.headerLength);
- EXPECT_TRUE(rtp_header.extension.hasAudioLevel);
- EXPECT_TRUE(rtp_header.extension.voiceActivity);
- EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
- EXPECT_EQ(1u, rtp_header.numCSRCs);
- EXPECT_EQ(csrcs[0], rtp_header.arrOfCSRCs[0]);
-}
-
-TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
- EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
- EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
- 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));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransportSequenceNumber,
- kTransportSequenceNumberExtensionId));
-
- size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), 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);
- map.Register(kRtpExtensionTransportSequenceNumber,
- kTransportSequenceNumberExtensionId);
- 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_TRUE(rtp_header.extension.hasTransportSequenceNumber);
- EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset);
- EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime);
- EXPECT_TRUE(rtp_header.extension.voiceActivity);
- EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
- EXPECT_EQ(kTransportSequenceNumber,
- rtp_header.extension.transportSequenceNumber);
-
- // Parse without map extension
- webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
-
- 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_FALSE(rtp_header2.extension.hasTransportSequenceNumber);
-
- EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
- EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
- EXPECT_FALSE(rtp_header2.extension.voiceActivity);
- EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
- EXPECT_EQ(0u, rtp_header2.extension.transportSequenceNumber);
-}
-
TEST_F(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, &mock_paced_sender_,
@@ -908,14 +583,13 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
- int rtp_length_int = rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);
- ASSERT_NE(-1, rtp_length_int);
- size_t rtp_length = static_cast<size_t>(rtp_length_int);
+ auto packet =
+ BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, capture_time_ms);
+ size_t packet_size = packet->size();
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- capture_time_ms, kAllowRetransmission,
+ EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
+ kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_EQ(0, transport_.packets_sent_);
@@ -928,10 +602,10 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
// Process send bucket. Packet should now be sent.
EXPECT_EQ(1, transport_.packets_sent_);
- EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
+ EXPECT_EQ(packet_size, transport_.last_sent_packet_len_);
// Parse sent packet.
webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
- rtp_length);
+ packet_size);
webrtc::RTPHeader rtp_header;
RtpHeaderExtensionMap map;
map.Register(kRtpExtensionTransmissionTimeOffset,
@@ -961,14 +635,13 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
- int rtp_length_int = rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);
- ASSERT_NE(-1, rtp_length_int);
- size_t rtp_length = static_cast<size_t>(rtp_length_int);
+ auto packet =
+ BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, capture_time_ms);
+ size_t packet_size = packet->size();
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- capture_time_ms, kAllowRetransmission,
+ EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
+ kAllowRetransmission,
RtpPacketSender::kNormalPriority));
EXPECT_EQ(0, transport_.packets_sent_);
@@ -979,7 +652,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
const int kStoredTimeInMs = 100;
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
- EXPECT_EQ(rtp_length_int, rtp_sender_->ReSendPacket(kSeqNum));
+ EXPECT_EQ(static_cast<int>(packet_size), rtp_sender_->ReSendPacket(kSeqNum));
EXPECT_EQ(0, transport_.packets_sent_);
rtp_sender_->TimeToSendPacket(kSeqNum, capture_time_ms, false,
@@ -987,11 +660,11 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
// Process send bucket. Packet should now be sent.
EXPECT_EQ(1, transport_.packets_sent_);
- EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
+ EXPECT_EQ(packet_size, transport_.last_sent_packet_len_);
// Parse sent packet.
webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
- rtp_length);
+ packet_size);
webrtc::RTPHeader rtp_header;
RtpHeaderExtensionMap map;
map.Register(kRtpExtensionTransmissionTimeOffset,
@@ -1042,15 +715,14 @@ TEST_F(RtpSenderTest, SendPadding) {
webrtc::RTPHeader rtp_header;
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
- int rtp_length_int = rtp_sender_->BuildRtpHeader(
- packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
+ auto packet =
+ BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
const uint32_t media_packet_timestamp = timestamp;
- ASSERT_NE(-1, rtp_length_int);
- size_t rtp_length = static_cast<size_t>(rtp_length_int);
+ size_t packet_size = packet->size();
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- capture_time_ms, kAllowRetransmission,
+ EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
+ kAllowRetransmission,
RtpPacketSender::kNormalPriority));
int total_packets_sent = 0;
@@ -1100,27 +772,25 @@ 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);
- ASSERT_NE(-1, rtp_length_int);
- rtp_length = static_cast<size_t>(rtp_length_int);
+ packet = BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
+ packet_size = packet->size();
EXPECT_CALL(mock_paced_sender_,
InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _));
// Packet should be stored in a send bucket.
- EXPECT_TRUE(rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- capture_time_ms, kAllowRetransmission,
+ EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
+ kAllowRetransmission,
RtpPacketSender::kNormalPriority));
rtp_sender_->TimeToSendPacket(seq_num, capture_time_ms, false,
PacketInfo::kNotAProbe);
// Process send bucket.
EXPECT_EQ(++total_packets_sent, transport_.packets_sent_);
- EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
+ EXPECT_EQ(packet_size, 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_, packet_size,
+ &rtp_header));
// Verify sequence number and timestamp.
EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
@@ -1759,6 +1429,7 @@ TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
// Verify that all packets of a frame have CVO byte set.
TEST_F(RtpSenderVideoTest, SendVideoWithCVO) {
+ uint8_t kFrame[kMaxPacketLength];
RTPVideoHeader hdr = {0};
hdr.rotation = kVideoRotation_90;
@@ -1771,7 +1442,7 @@ TEST_F(RtpSenderVideoTest, SendVideoWithCVO) {
rtp_sender_->RtpHeaderExtensionLength());
rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey, kPayload,
- kTimestamp, 0, packet_, sizeof(packet_), nullptr,
+ kTimestamp, 0, kFrame, sizeof(kFrame), nullptr,
&hdr);
RtpHeaderExtensionMap map;
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