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

Issue 1539423003: [rtp_rtcp] Lint errors cleaned from rtp_utility (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: 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 1ca7831ab2cb9c153cae70c4f2d204a5f759efdf..6bc122201adfd64881afffed7b28dae40fa29990 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -208,7 +208,7 @@ class RtpSenderVideoTest : public RtpSenderTest {
} else {
ASSERT_EQ(kRtpHeaderSize, length);
}
- ASSERT_TRUE(rtp_parser.Parse(rtp_header, map));
+ ASSERT_TRUE(rtp_parser.Parse(&rtp_header, map));
ASSERT_FALSE(rtp_parser.RTCP());
EXPECT_EQ(payload_, rtp_header.payloadType);
EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
@@ -335,7 +335,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) {
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
webrtc::RTPHeader rtp_header;
- const bool valid_rtp_header = rtp_parser.Parse(rtp_header, nullptr);
+ const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, nullptr);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
@@ -370,7 +370,7 @@ TEST_F(RtpSenderTestWithoutPacer,
RtpHeaderExtensionMap map;
map.Register(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
- const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+ const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
@@ -381,7 +381,7 @@ TEST_F(RtpSenderTestWithoutPacer,
// Parse without map extension
webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+ const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
@@ -410,7 +410,7 @@ TEST_F(RtpSenderTestWithoutPacer,
RtpHeaderExtensionMap map;
map.Register(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
- const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+ const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
@@ -437,7 +437,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
RtpHeaderExtensionMap map;
map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
- const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+ const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
@@ -448,7 +448,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
// Parse without map extension
webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+ const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
@@ -476,7 +476,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
webrtc::RTPHeader rtp_header;
- ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map));
+ ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header);
EXPECT_EQ(length, rtp_header.headerLength);
@@ -504,7 +504,7 @@ TEST_F(RtpSenderTestWithoutPacer,
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
webrtc::RTPHeader rtp_header;
- ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map));
+ ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header, false);
EXPECT_EQ(length, rtp_header.headerLength);
@@ -525,12 +525,12 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
webrtc::RTPHeader rtp_header;
// Updating audio level is done in RTPSenderAudio, so simulate it here.
- rtp_parser.Parse(rtp_header);
+ 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);
+ const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
@@ -542,7 +542,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
// Parse without map extension
webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+ const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
@@ -579,7 +579,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
webrtc::RTPHeader rtp_header;
// Updating audio level is done in RTPSenderAudio, so simulate it here.
- rtp_parser.Parse(rtp_header);
+ rtp_parser.Parse(&rtp_header);
rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
RtpHeaderExtensionMap map;
@@ -589,7 +589,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
map.Register(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
- const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+ const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
@@ -608,7 +608,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
// Parse without map extension
webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
+ const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
@@ -667,7 +667,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
map.Register(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
- const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+ const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
// Verify transmission time offset.
@@ -727,7 +727,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
map.Register(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
- const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+ const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
// Verify transmission time offset.
@@ -934,7 +934,7 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
webrtc::RTPHeader rtp_header;
- ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+ ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
const uint8_t* payload_data =
GetPayloadData(rtp_header, transport_.last_sent_packet_);
@@ -959,7 +959,7 @@ TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
- ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+ ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_);
generic_header = *payload_data++;
@@ -1217,7 +1217,7 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
webrtc::RTPHeader rtp_header;
- ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+ ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
const uint8_t* payload_data =
GetPayloadData(rtp_header, transport_.last_sent_packet_);
@@ -1246,7 +1246,7 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
webrtc::RTPHeader rtp_header;
- ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+ ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
const uint8_t* payload_data =
GetPayloadData(rtp_header, transport_.last_sent_packet_);
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