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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: addressed feedback Created 4 years, 6 months ago
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« webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc ('K') | « webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.h » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('J')
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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 f0b6411af2711ffc3d2d39c446774268e5ecebec..86efa35ef013bdb2e77b2da3aea49eada639ac88 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -182,7 +182,7 @@ 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);
@@ -203,7 +203,7 @@ class RtpSenderTest : public ::testing::Test {
EXPECT_EQ(0, rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs,
- kPayload, sizeof(kPayload), nullptr));
+ kPayload, sizeof(kPayload), nullptr, nullptr));
}
};
@@ -233,7 +233,7 @@ 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(),
@@ -360,7 +360,7 @@ 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);
@@ -391,7 +391,7 @@ 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);
@@ -430,7 +430,7 @@ 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);
@@ -457,7 +457,7 @@ 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);
@@ -542,7 +542,7 @@ 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
@@ -570,7 +570,7 @@ 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
@@ -588,7 +588,7 @@ 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);
@@ -631,7 +631,7 @@ 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
@@ -675,7 +675,7 @@ 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);
@@ -745,7 +745,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
kAbsoluteSendTimeExtensionId));
rtp_sender_->SetTargetBitrate(300000);
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);
@@ -799,7 +799,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
kAbsoluteSendTimeExtensionId));
rtp_sender_->SetTargetBitrate(300000);
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) {
rtp_sender_->SetTargetBitrate(300000);
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);
@@ -1110,9 +1110,9 @@ 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_);
@@ -1136,9 +1136,9 @@ 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_);
@@ -1189,18 +1189,18 @@ 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_);
@@ -1254,9 +1254,9 @@ 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);
}
@@ -1334,9 +1334,9 @@ 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;
@@ -1376,9 +1376,9 @@ 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;
@@ -1395,9 +1395,9 @@ 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_);
@@ -1424,9 +1424,9 @@ 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_);
@@ -1477,13 +1477,13 @@ 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);
@@ -1495,7 +1495,7 @@ 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.
@@ -1514,9 +1514,9 @@ 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);
« webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc ('K') | « webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.h » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('J')

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