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

Issue 1512493002: [rtp_rtcp] lint whitespace warning removed from most source/ files (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years 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 332a0f81f9e098f30b16168a630a8336fc1e9296..dcae547b1c42cf364ba6dca31ab337cf1ecce972 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -90,9 +90,7 @@ class LoopbackTransportTest : public webrtc::Transport {
sent_packets_.push_back(buffer);
return true;
}
- bool SendRtcp(const uint8_t* data, size_t len) override {
- return false;
- }
+ bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
int packets_sent_;
size_t last_sent_packet_len_;
size_t total_bytes_sent_;
@@ -163,11 +161,8 @@ class RtpSenderTest : public ::testing::Test {
void SendPacket(int64_t capture_time_ms, int payload_length) {
uint32_t timestamp = capture_time_ms * 90;
- int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
- kPayload,
- kMarkerBit,
- timestamp,
- capture_time_ms);
+ int32_t rtp_length = rtp_sender_->BuildRTPheader(
+ packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
ASSERT_GE(rtp_length, 0);
// Packet should be stored in a send bucket.
@@ -186,7 +181,7 @@ class RtpSenderTestWithoutPacer : public RtpSenderTest {
class RtpSenderVideoTest : public RtpSenderTest {
protected:
- virtual void SetUp() override {
+ void SetUp() override {
// TODO(pbos): Set up to use pacer.
SetUpRtpSender(false);
rtp_sender_video_.reset(
@@ -228,53 +223,57 @@ TEST_F(RtpSenderTestWithoutPacer,
RegisterRtpTransmissionTimeOffsetHeaderExtension) {
EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+ kRtpExtensionTransmissionTimeOffset,
+ kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength,
rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset));
+ kRtpExtensionTransmissionTimeOffset));
EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
}
TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAbsoluteSendTimeHeaderExtension) {
EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+ EXPECT_EQ(
+ 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsoluteSendTimeExtensionId));
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kAbsoluteSendTimeLength),
rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime));
+ kRtpExtensionAbsoluteSendTime));
EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
}
TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAudioLevelHeaderExtension) {
EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+ EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+ kAudioLevelExtensionId));
EXPECT_EQ(
RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAudioLevelLength),
rtp_sender_->RtpHeaderExtensionTotalLength());
- EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
- kRtpExtensionAudioLevel));
+ EXPECT_EQ(0,
+ rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel));
EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
}
TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) {
EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+ kRtpExtensionTransmissionTimeOffset,
+ kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kTransmissionTimeOffsetLength),
rtp_sender_->RtpHeaderExtensionTotalLength());
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+ EXPECT_EQ(
+ 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsoluteSendTimeExtensionId));
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kTransmissionTimeOffsetLength +
kAbsoluteSendTimeLength),
rtp_sender_->RtpHeaderExtensionTotalLength());
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+ EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+ kAudioLevelExtensionId));
EXPECT_EQ(RtpUtility::Word32Align(
kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength +
kAbsoluteSendTimeLength + kAudioLevelLength),
@@ -290,18 +289,18 @@ TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) {
// Deregister starts.
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset));
+ kRtpExtensionTransmissionTimeOffset));
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kAbsoluteSendTimeLength +
kAudioLevelLength + kVideoRotationLength),
rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime));
+ kRtpExtensionAbsoluteSendTime));
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kAudioLevelLength + kVideoRotationLength),
rtp_sender_->RtpHeaderExtensionTotalLength());
- EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
- kRtpExtensionAudioLevel));
+ EXPECT_EQ(0,
+ rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel));
EXPECT_EQ(
RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength),
rtp_sender_->RtpHeaderExtensionTotalLength());
@@ -354,7 +353,8 @@ TEST_F(RtpSenderTestWithoutPacer,
BuildRTPPacketWithTransmissionOffsetExtension) {
EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+ kRtpExtensionTransmissionTimeOffset,
+ kTransmissionTimeOffsetExtensionId));
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -393,7 +393,8 @@ TEST_F(RtpSenderTestWithoutPacer,
const int kNegTimeOffset = -500;
EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kNegTimeOffset));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+ kRtpExtensionTransmissionTimeOffset,
+ kTransmissionTimeOffsetExtensionId));
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -419,8 +420,9 @@ TEST_F(RtpSenderTestWithoutPacer,
TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+ EXPECT_EQ(
+ 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsoluteSendTimeExtensionId));
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -508,8 +510,8 @@ TEST_F(RtpSenderTestWithoutPacer,
}
TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+ EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+ kAudioLevelExtensionId));
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -554,11 +556,13 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
EXPECT_EQ(0,
rtp_sender_->SetTransportSequenceNumber(kTransportSequenceNumber));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+ kRtpExtensionTransmissionTimeOffset,
+ kTransmissionTimeOffsetExtensionId));
+ EXPECT_EQ(
+ 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsoluteSendTimeExtensionId));
+ EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+ kAudioLevelExtensionId));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
@@ -626,9 +630,11 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+ kRtpExtensionTransmissionTimeOffset,
+ kTransmissionTimeOffsetExtensionId));
+ EXPECT_EQ(
+ 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsoluteSendTimeExtensionId));
rtp_sender_->SetTargetBitrate(300000);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
int rtp_length_int = rtp_sender_->BuildRTPheader(
@@ -676,9 +682,11 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+ kRtpExtensionTransmissionTimeOffset,
+ kTransmissionTimeOffsetExtensionId));
+ EXPECT_EQ(
+ 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsoluteSendTimeExtensionId));
rtp_sender_->SetTargetBitrate(300000);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
int rtp_length_int = rtp_sender_->BuildRTPheader(
@@ -740,10 +748,12 @@ TEST_F(RtpSenderTest, SendPadding) {
rtp_sender_->SetStorePacketsStatus(true, 10);
size_t rtp_header_len = kRtpHeaderSize;
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+ kRtpExtensionTransmissionTimeOffset,
+ kTransmissionTimeOffsetExtensionId));
rtp_header_len += 4; // 4 bytes extension.
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+ EXPECT_EQ(
+ 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsoluteSendTimeExtensionId));
rtp_header_len += 4; // 4 bytes extension.
rtp_header_len += 4; // 4 extra bytes common to all extension headers.
@@ -815,8 +825,8 @@ TEST_F(RtpSenderTest, SendPadding) {
// Send a regular video packet again.
capture_time_ms = fake_clock_.TimeInMilliseconds();
- rtp_length_int = rtp_sender_->BuildRTPheader(
- packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
+ rtp_length_int = rtp_sender_->BuildRTPheader(packet_, kPayload, kMarkerBit,
+ timestamp, capture_time_ms);
ASSERT_NE(-1, rtp_length_int);
rtp_length = static_cast<size_t>(rtp_length_int);
@@ -830,8 +840,8 @@ TEST_F(RtpSenderTest, SendPadding) {
EXPECT_EQ(++total_packets_sent, transport_.packets_sent_);
EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
// Parse sent packet.
- ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, rtp_length,
- &rtp_header));
+ ASSERT_TRUE(
+ rtp_parser->Parse(transport_.last_sent_packet_, rtp_length, &rtp_header));
// Verify sequence number and timestamp.
EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
@@ -858,8 +868,9 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
uint16_t seq_num = kSeqNum;
rtp_sender_->SetStorePacketsStatus(true, 10);
int32_t rtp_header_len = kRtpHeaderSize;
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+ EXPECT_EQ(
+ 0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsoluteSendTimeExtensionId));
rtp_header_len += 4; // 4 bytes extension.
rtp_header_len += 4; // 4 extra bytes common to all extension headers.
@@ -876,8 +887,8 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
kAbsoluteSendTimeExtensionId);
rtp_sender_->SetTargetBitrate(300000);
const size_t kNumPayloadSizes = 10;
- const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, 750,
- 800, 850, 900, 950};
+ const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700,
+ 750, 800, 850, 900, 950};
// Send 10 packets of increasing size.
for (size_t i = 0; i < kNumPayloadSizes; ++i) {
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
@@ -923,8 +934,8 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
webrtc::RTPHeader rtp_header;
ASSERT_TRUE(rtp_parser.Parse(rtp_header));
- const uint8_t* payload_data = GetPayloadData(rtp_header,
- transport_.last_sent_packet_);
+ const uint8_t* payload_data =
+ GetPayloadData(rtp_header, transport_.last_sent_packet_);
uint8_t generic_header = *payload_data++;
ASSERT_EQ(sizeof(payload) + sizeof(generic_header),
@@ -1043,9 +1054,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
- ASSERT_EQ(
- 0,
- rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 0, 1500));
+ ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
+ 0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
@@ -1057,13 +1067,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
// Send a few frames.
for (uint32_t i = 0; i < kNumPackets; ++i) {
ASSERT_EQ(0,
- rtp_sender_->SendOutgoingData(kVideoFrameKey,
- payload_type,
- 1234,
- 4321,
- payload,
- sizeof(payload),
- 0));
+ rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+ 4321, payload, sizeof(payload), 0));
fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
}
@@ -1100,8 +1105,7 @@ class RtpSenderAudioTest : public RtpSenderTest {
TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
class TestCallback : public StreamDataCountersCallback {
public:
- TestCallback()
- : StreamDataCountersCallback(), ssrc_(0), counters_() {}
+ TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
virtual ~TestCallback() {}
void DataCountersUpdated(const StreamDataCounters& counters,
@@ -1127,7 +1131,6 @@ TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
MatchPacketCounter(counters.retransmitted, counters_.retransmitted);
EXPECT_EQ(counters.fec.packets, counters_.fec.packets);
}
-
} callback;
const uint8_t kRedPayloadType = 96;
@@ -1214,8 +1217,8 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
webrtc::RTPHeader rtp_header;
ASSERT_TRUE(rtp_parser.Parse(rtp_header));
- const uint8_t* payload_data = GetPayloadData(rtp_header,
- transport_.last_sent_packet_);
+ const uint8_t* payload_data =
+ GetPayloadData(rtp_header, transport_.last_sent_packet_);
ASSERT_EQ(sizeof(payload),
GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
@@ -1225,8 +1228,8 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
- kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+ EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+ kAudioLevelExtensionId));
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
const uint8_t payload_type = 127;
@@ -1243,19 +1246,20 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
webrtc::RTPHeader rtp_header;
ASSERT_TRUE(rtp_parser.Parse(rtp_header));
- const uint8_t* payload_data = GetPayloadData(rtp_header,
- transport_.last_sent_packet_);
+ const uint8_t* payload_data =
+ GetPayloadData(rtp_header, transport_.last_sent_packet_);
ASSERT_EQ(sizeof(payload),
GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
- uint8_t extension[] = { 0xbe, 0xde, 0x00, 0x01,
- (kAudioLevelExtensionId << 4) + 0, // ID + length.
- kAudioLevel, // Data.
- 0x00, 0x00 // Padding.
- };
+ uint8_t extension[] = {
+ 0xbe, 0xde, 0x00, 0x01,
+ (kAudioLevelExtensionId << 4) + 0, // ID + length.
+ kAudioLevel, // Data.
+ 0x00, 0x00 // Padding.
+ };
EXPECT_EQ(0, memcmp(extension, payload_data - sizeof(extension),
sizeof(extension)));
@@ -1270,14 +1274,14 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event";
uint8_t payload_type = 126;
- ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 0,
- 0, 0));
+ ASSERT_EQ(0,
+ rtp_sender_->RegisterPayload(payload_name, payload_type, 0, 0, 0));
// For Telephone events, payload is not added to the registered payload list,
// it will register only the payload used for audio stream.
// Registering the payload again for audio stream with different payload name.
strcpy(payload_name, "payload_name");
- ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 8000,
- 1, 0));
+ 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);
@@ -1298,8 +1302,7 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
ASSERT_TRUE(rtp_parser.get() != nullptr);
webrtc::RTPHeader rtp_header;
ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
- transport_.last_sent_packet_len_,
- &rtp_header));
+ transport_.last_sent_packet_len_, &rtp_header));
// Marker Bit should be set to 1 for first packet.
EXPECT_TRUE(rtp_header.markerBit);
@@ -1307,8 +1310,7 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
capture_time_ms + 4000, 0, nullptr,
0, nullptr));
ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
- transport_.last_sent_packet_len_,
- &rtp_header));
+ transport_.last_sent_packet_len_, &rtp_header));
// Marker Bit should be set to 0 for rest of the packets.
EXPECT_FALSE(rtp_header.markerBit);
}
@@ -1321,19 +1323,13 @@ TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
- ASSERT_EQ(
- 0,
- rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000, 0, 1500));
+ ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000,
+ 0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_EQ(0,
- rtp_sender_->SendOutgoingData(kVideoFrameKey,
- kPayloadType,
- 1234,
- 4321,
- payload,
- sizeof(payload),
- 0));
+ ASSERT_EQ(
+ 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234, 4321,
+ payload, sizeof(payload), 0));
// Will send 2 full-size padding packets.
rtp_sender_->TimeToSendPadding(1);
@@ -1353,17 +1349,17 @@ TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
EXPECT_EQ(rtx_stats.transmitted.padding_bytes, 2 * kMaxPaddingSize);
EXPECT_EQ(rtp_stats.transmitted.TotalBytes(),
- rtp_stats.transmitted.payload_bytes +
- rtp_stats.transmitted.header_bytes +
- rtp_stats.transmitted.padding_bytes);
+ rtp_stats.transmitted.payload_bytes +
+ rtp_stats.transmitted.header_bytes +
+ rtp_stats.transmitted.padding_bytes);
EXPECT_EQ(rtx_stats.transmitted.TotalBytes(),
- rtx_stats.transmitted.payload_bytes +
- rtx_stats.transmitted.header_bytes +
- rtx_stats.transmitted.padding_bytes);
+ rtx_stats.transmitted.payload_bytes +
+ rtx_stats.transmitted.header_bytes +
+ rtx_stats.transmitted.padding_bytes);
- EXPECT_EQ(transport_.total_bytes_sent_,
- rtp_stats.transmitted.TotalBytes() +
- rtx_stats.transmitted.TotalBytes());
+ EXPECT_EQ(
+ transport_.total_bytes_sent_,
+ rtp_stats.transmitted.TotalBytes() + rtx_stats.transmitted.TotalBytes());
}
TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
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