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Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.

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- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.

This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.

BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org
2014-03-06 23:49:08 +00:00
parent 79047f99c1
commit ebdb0e3ad0
18 changed files with 490 additions and 300 deletions
Download Patch File Download Diff File Expand all files Collapse all files

155
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -160,11 +160,8 @@ TEST_F(RtpSenderTest, RegisterRtpAudioLevelHeaderExtension) {
EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, kAudioLevelExtensionId));
// Accounted size for audio level is zero because it is currently specially
// treated by RTPSenderAudio.
EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
// EXPECT_EQ(kRtpOneByteHeaderLength + kAudioLevelLength,
// rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(kRtpOneByteHeaderLength + kAudioLevelLength,
rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
kRtpExtensionAudioLevel));
EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
@ -183,14 +180,16 @@ TEST_F(RtpSenderTest, RegisterRtpHeaderExtensions) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, kAudioLevelExtensionId));
EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength +
kAbsoluteSendTimeLength, rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset));
EXPECT_EQ(kRtpOneByteHeaderLength + kAbsoluteSendTimeLength,
kAbsoluteSendTimeLength + kAudioLevelLength,
rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset));
EXPECT_EQ(kRtpOneByteHeaderLength + kAbsoluteSendTimeLength +
kAudioLevelLength, rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime));
EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(kRtpOneByteHeaderLength + kAudioLevelLength,
rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
kRtpExtensionAudioLevel));
EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
@ -202,23 +201,24 @@ TEST_F(RtpSenderTest, BuildRTPPacket) {
kMarkerBit,
kTimestamp,
0);
EXPECT_EQ(12, length);
EXPECT_EQ(kRtpHeaderSize, length);
// Verify
webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
webrtc::RTPHeader rtp_header;
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, NULL);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header);
EXPECT_EQ(length, rtp_header.headerLength);
EXPECT_FALSE(rtp_header.extension.hasTransmissionTimeOffset);
EXPECT_FALSE(rtp_header.extension.hasAbsoluteSendTime);
EXPECT_FALSE(rtp_header.extension.hasAudioLevel);
EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
EXPECT_EQ(0u, rtp_header.extension.absoluteSendTime);
EXPECT_EQ(0u, rtp_header.extension.audioLevel);
}
TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
@ -231,7 +231,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
kMarkerBit,
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
length);
// Verify
webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
@ -246,6 +247,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header);
EXPECT_EQ(length, rtp_header.headerLength);
EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset);
// Parse without map extension
@ -255,6 +257,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
EXPECT_EQ(length, rtp_header2.headerLength);
EXPECT_FALSE(rtp_header2.extension.hasTransmissionTimeOffset);
EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
}
@ -269,7 +272,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
kMarkerBit,
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
length);
// Verify
webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
@ -284,6 +288,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header);
EXPECT_EQ(length, rtp_header.headerLength);
EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
EXPECT_EQ(kNegTimeOffset, rtp_header.extension.transmissionTimeOffset);
}
@ -297,7 +302,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
kMarkerBit,
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
length);
// Verify
webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
@ -311,6 +317,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header);
EXPECT_EQ(length, rtp_header.headerLength);
EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime);
// Parse without map extension
@ -320,9 +327,54 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
EXPECT_EQ(length, rtp_header2.headerLength);
EXPECT_FALSE(rtp_header2.extension.hasAbsoluteSendTime);
EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
}
TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, kAudioLevelExtensionId));
int32_t length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
kTimestamp,
0);
EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
length);
// Verify
webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
webrtc::RTPHeader rtp_header;
// Updating audio level is done in RTPSenderAudio, so simulate it here.
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);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header);
EXPECT_EQ(length, rtp_header.headerLength);
EXPECT_TRUE(rtp_header.extension.hasAudioLevel);
// Expect kAudioLevel + 0x80 because we set "voiced" to true in the call to
// UpdateAudioLevel(), above.
EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel);
// Parse without map extension
webrtc::RTPHeader rtp_header2;
const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
EXPECT_EQ(length, rtp_header2.headerLength);
EXPECT_FALSE(rtp_header2.extension.hasAudioLevel);
EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
}
TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
@ -330,30 +382,42 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, kAudioLevelExtensionId));
int32_t length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
EXPECT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
length);
// Verify
webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
webrtc::RTPHeader rtp_header;
// Updating audio level is done in RTPSenderAudio, so simulate it here.
rtp_parser.Parse(rtp_header);
rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
RtpHeaderExtensionMap map;
map.Register(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header);
EXPECT_EQ(length, rtp_header.headerLength);
EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
EXPECT_TRUE(rtp_header.extension.hasAbsoluteSendTime);
EXPECT_TRUE(rtp_header.extension.hasAudioLevel);
EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset);
EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime);
EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel);
// Parse without map extension
webrtc::RTPHeader rtp_header2;
@ -362,8 +426,12 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
EXPECT_EQ(length, rtp_header2.headerLength);
EXPECT_FALSE(rtp_header2.extension.hasTransmissionTimeOffset);
EXPECT_FALSE(rtp_header2.extension.hasAbsoluteSendTime);
EXPECT_FALSE(rtp_header2.extension.hasAudioLevel);
EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
}
TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
@ -493,7 +561,7 @@ TEST_F(RtpSenderTest, SendPadding) {
uint16_t seq_num = kSeqNum;
uint32_t timestamp = kTimestamp;
rtp_sender_->SetStorePacketsStatus(true, 10);
int rtp_header_len = 12;
int32_t rtp_header_len = kRtpHeaderSize;
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
rtp_header_len += 4; // 4 bytes extension.
@ -613,7 +681,7 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
uint16_t seq_num = kSeqNum;
rtp_sender_->SetStorePacketsStatus(true, 10);
int rtp_header_len = 12;
int32_t rtp_header_len = kRtpHeaderSize;
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
rtp_header_len += 4; // 4 bytes extension.
@ -939,48 +1007,6 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
rtp_sender_->RegisterRtpStatisticsCallback(NULL);
}
TEST_F(RtpSenderAudioTest, BuildRTPPacketWithAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(true,
kAudioLevelExtensionId));
EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, kAudioLevelExtensionId));
int32_t length = rtp_sender_->BuildRTPheader(packet_,
kAudioPayload,
kMarkerBit,
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
// Currently, no space is added by for header extension by BuildRTPHeader().
EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
// Verify
webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
webrtc::RTPHeader rtp_header;
RtpHeaderExtensionMap map;
map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header);
EXPECT_EQ(length, rtp_header.headerLength);
// TODO(solenberg): Should verify that we got audio level in header extension.
// Parse without map extension
webrtc::RTPHeader rtp_header2;
const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
EXPECT_EQ(length, rtp_header2.headerLength);
// TODO(solenberg): Should verify that we didn't get audio level.
EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(false, 0));
}
TEST_F(RtpSenderAudioTest, SendAudio) {
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
const uint8_t payload_type = 127;
@ -1007,8 +1033,6 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
}
TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(true,
kAudioLevelExtensionId));
EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, kAudioLevelExtensionId));
@ -1044,7 +1068,6 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
EXPECT_EQ(0, memcmp(extension, payload_data - sizeof(extension),
sizeof(extension)));
EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(false, 0));
}
} // namespace webrtc