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

Issue 2067673004: Style cleanups in RtpSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 4 years, 5 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 e2709db31c4b401b178e059270bf00bac7a2ac26..302f6ddb4d82bb225a231b1497563d7a342dec07 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -185,7 +185,7 @@ 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(
+ int32_t rtp_length = rtp_sender_->BuildRtpHeader(
packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
ASSERT_GE(rtp_length, 0);
@@ -206,7 +206,7 @@ class RtpSenderTest : public ::testing::Test {
EXPECT_EQ(0, rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs,
- kPayload, sizeof(kPayload), nullptr));
+ kPayload, sizeof(kPayload), nullptr, nullptr));
}
};
@@ -236,7 +236,7 @@ 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(
+ 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(),
@@ -363,7 +363,7 @@ TEST_F(RtpSenderTestWithoutPacer, RegisterRtpVideoRotationHeaderExtension) {
}
TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) {
- size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+ size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
ASSERT_EQ(kRtpHeaderSize, length);
@@ -394,7 +394,7 @@ TEST_F(RtpSenderTestWithoutPacer,
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
- size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+ size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
@@ -433,7 +433,7 @@ TEST_F(RtpSenderTestWithoutPacer,
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
- size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+ size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
@@ -460,7 +460,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
- size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+ size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
@@ -545,7 +545,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
map.Register(kRtpExtensionVideoRotation, kVideoRotationExtensionId);
size_t length = static_cast<size_t>(
- rtp_sender_->BuildRTPheader(packet_, kPayload, true, kTimestamp, 0));
+ rtp_sender_->BuildRtpHeader(packet_, kPayload, true, kTimestamp, 0));
ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
// Verify
@@ -573,7 +573,7 @@ TEST_F(RtpSenderTestWithoutPacer,
map.Register(kRtpExtensionVideoRotation, kVideoRotationExtensionId);
size_t length = static_cast<size_t>(
- rtp_sender_->BuildRTPheader(packet_, kPayload, false, kTimestamp, 0));
+ rtp_sender_->BuildRtpHeader(packet_, kPayload, false, kTimestamp, 0));
ASSERT_EQ(kRtpHeaderSize, length);
// Verify
@@ -591,7 +591,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
kAudioLevelExtensionId));
- size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+ size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
@@ -634,7 +634,7 @@ TEST_F(RtpSenderTestWithoutPacer,
std::vector<uint32_t> csrcs;
csrcs.push_back(0x23456789);
rtp_sender_->SetCsrcs(csrcs);
- size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+ size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
// Verify
@@ -678,7 +678,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
- size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+ size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
@@ -747,7 +747,7 @@ 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(
+ 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);
@@ -800,7 +800,7 @@ 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(
+ 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);
@@ -881,7 +881,7 @@ TEST_F(RtpSenderTest, SendPadding) {
webrtc::RTPHeader rtp_header;
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
- int rtp_length_int = rtp_sender_->BuildRTPheader(
+ int rtp_length_int = rtp_sender_->BuildRtpHeader(
packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
const uint32_t media_packet_timestamp = timestamp;
ASSERT_NE(-1, rtp_length_int);
@@ -939,7 +939,7 @@ 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,
+ 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);
@@ -1114,9 +1114,9 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
uint8_t payload[] = {47, 11, 32, 93, 89};
// Send keyframe
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1140,9 +1140,9 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
payload[1] = 42;
payload[4] = 13;
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
- 1234, 4321, payload,
- sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameDelta, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1193,18 +1193,18 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
.Times(::testing::AtLeast(2));
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
EXPECT_EQ(1U, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
EXPECT_EQ(1, callback.frame_counts_.key_frames);
EXPECT_EQ(0, callback.frame_counts_.delta_frames);
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
- 1234, 4321, payload,
- sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameDelta, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr));
EXPECT_EQ(2U, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
@@ -1266,9 +1266,9 @@ 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));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameKey, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr));
fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
}
@@ -1347,9 +1347,9 @@ TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
rtp_sender_->RegisterRtpStatisticsCallback(&callback);
// Send a frame.
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
StreamDataCounters expected;
expected.transmitted.payload_bytes = 6;
expected.transmitted.header_bytes = 12;
@@ -1389,9 +1389,9 @@ TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
fec_params.fec_rate = 1;
fec_params.max_fec_frames = 1;
rtp_sender_->SetFecParameters(&fec_params, &fec_params);
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
- 1234, 4321, payload,
- sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+ kVideoFrameDelta, payload_type, 1234, 4321, payload,
+ sizeof(payload), nullptr, nullptr));
expected.transmitted.payload_bytes = 40;
expected.transmitted.header_bytes = 60;
expected.transmitted.packets = 5;
@@ -1408,9 +1408,9 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1437,9 +1437,9 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_EQ(
- 0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
- payload, sizeof(payload), nullptr));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+ 4321, payload, sizeof(payload),
+ nullptr, nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1490,13 +1490,13 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
// timestamp. So for first call it will skip since the duration is zero.
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms, 0, nullptr, 0,
- nullptr));
+ nullptr, nullptr));
// 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(kEmptyFrame, payload_type,
capture_time_ms + 2000, 0, nullptr,
- 0, nullptr));
+ 0, nullptr, nullptr));
std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
webrtc::RtpHeaderParser::Create());
ASSERT_TRUE(rtp_parser.get() != nullptr);
@@ -1508,7 +1508,7 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
capture_time_ms + 4000, 0, nullptr,
- 0, nullptr));
+ 0, nullptr, nullptr));
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.
@@ -1527,9 +1527,9 @@ TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
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),
+ nullptr, nullptr));
// Will send 2 full-size padding packets.
rtp_sender_->TimeToSendPadding(1, PacketInfo::kNotAProbe);
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