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

Issue 2007743003: Add sender controlled playout delay limits (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@cleanup_rtp_hdr_extensions
Patch Set: Rename OnReceivedRtcpReport to OnReceivedRtcpReportBlocks Created 4 years, 6 months ago
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« webrtc/modules/modules.gyp ('K') | « webrtc/modules/rtp_rtcp/source/rtp_sender.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('j') | no next file with comments »
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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 aa882d2baac610fccf00a640fff943a4f00323f8..f0b6411af2711ffc3d2d39c446774268e5ecebec 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -236,7 +236,7 @@ class RtpSenderVideoTest : public RtpSenderTest {
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_->RtpHeaderExtensionTotalLength(),
+ ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(),
length);
} else {
ASSERT_EQ(kRtpHeaderSize, length);
@@ -256,63 +256,63 @@ class RtpSenderVideoTest : public RtpSenderTest {
TEST_F(RtpSenderTestWithoutPacer,
RegisterRtpTransmissionTimeOffsetHeaderExtension) {
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength,
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset));
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
}
TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAbsoluteSendTimeHeaderExtension) {
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kAbsoluteSendTimeLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime));
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
}
TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAudioLevelHeaderExtension) {
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
kAudioLevelExtensionId));
EXPECT_EQ(
RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAudioLevelLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0,
rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel));
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
}
TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) {
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kTransmissionTimeOffsetLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kTransmissionTimeOffsetLength +
kAbsoluteSendTimeLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
kAudioLevelExtensionId));
EXPECT_EQ(RtpUtility::Word32Align(
kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength +
kAbsoluteSendTimeLength + kAudioLevelLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoRotation, kVideoRotationExtensionId));
EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
@@ -320,7 +320,7 @@ TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) {
kTransmissionTimeOffsetLength +
kAbsoluteSendTimeLength +
kAudioLevelLength + kVideoRotationLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
// Deregister starts.
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
@@ -328,35 +328,35 @@ TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) {
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kAbsoluteSendTimeLength +
kAudioLevelLength + kVideoRotationLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime));
EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
kAudioLevelLength + kVideoRotationLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0,
rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel));
EXPECT_EQ(
RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(
0, rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionVideoRotation));
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
}
TEST_F(RtpSenderTestWithoutPacer, RegisterRtpVideoRotationHeaderExtension) {
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoRotation, kVideoRotationExtensionId));
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
EXPECT_EQ(
RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
EXPECT_EQ(
0, rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionVideoRotation));
- EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
+ EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
}
TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) {
@@ -393,8 +393,7 @@ TEST_F(RtpSenderTestWithoutPacer,
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
- length);
+ ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
// Verify
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
@@ -433,8 +432,7 @@ TEST_F(RtpSenderTestWithoutPacer,
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
- length);
+ ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
// Verify
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
@@ -461,8 +459,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
- length);
+ ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
// Verify
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
@@ -546,8 +543,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
size_t length = static_cast<size_t>(
rtp_sender_->BuildRTPheader(packet_, kPayload, true, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
- length);
+ ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
// Verify
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
@@ -594,8 +590,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
- length);
+ ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
// Verify
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
@@ -682,8 +677,7 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
- ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
- length);
+ ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
// Verify
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
@@ -1595,7 +1589,7 @@ TEST_F(RtpSenderVideoTest, SendVideoWithCVO) {
EXPECT_EQ(
RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength),
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ rtp_sender_->RtpHeaderExtensionLength());
rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey, kPayload,
kTimestamp, 0, packet_, sizeof(packet_), nullptr,
« webrtc/modules/modules.gyp ('K') | « webrtc/modules/rtp_rtcp/source/rtp_sender.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('j') | no next file with comments »

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