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Diffstat (limited to 'chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc')
-rw-r--r--chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc165
1 files changed, 94 insertions, 71 deletions
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index ce615be0494..18482890f7a 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -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());
+ kAbsoluteSendTimeLength + kAudioLevelLength,
+ rtp_sender_->RtpHeaderExtensionTotalLength());
EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset));
- EXPECT_EQ(kRtpOneByteHeaderLength + kAbsoluteSendTimeLength,
- rtp_sender_->RtpHeaderExtensionTotalLength());
+ 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) {
@@ -376,7 +444,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
- rtp_sender_->SetTargetSendBitrate(300000);
+ rtp_sender_->SetTargetBitrate(300000);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
@@ -430,7 +498,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
- rtp_sender_->SetTargetSendBitrate(300000);
+ rtp_sender_->SetTargetBitrate(300000);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
@@ -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.
@@ -512,7 +580,7 @@ TEST_F(RtpSenderTest, SendPadding) {
kAbsoluteSendTimeExtensionId);
webrtc::RTPHeader rtp_header;
- rtp_sender_->SetTargetSendBitrate(300000);
+ rtp_sender_->SetTargetBitrate(300000);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
@@ -613,14 +681,14 @@ 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.
rtp_header_len += 4; // 4 extra bytes common to all extension headers.
- rtp_sender_->SetRTXStatus(kRtxRetransmitted | kRtxRedundantPayloads, true,
- 1234);
+ rtp_sender_->SetRTXStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+ rtp_sender_->SetRtxSsrc(1234);
// Create and set up parser.
scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
@@ -630,7 +698,7 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
kTransmissionTimeOffsetExtensionId);
rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId);
- rtp_sender_->SetTargetSendBitrate(300000);
+ rtp_sender_->SetTargetBitrate(300000);
const size_t kNumPayloadSizes = 10;
const int kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, 750,
800, 850, 900, 950};
@@ -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
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-07 06:33:14 +0000