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 ea6fb6adf3be5229cd08008a0a1c1f09b650c876..409be1a66ce16ad7fb9b6f5c4559eadf5f154ae3 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
@@ -320,6 +320,7 @@ TEST_F(RtpSenderTest, BuildRTPPacket) { |
EXPECT_FALSE(rtp_header.extension.hasAudioLevel); |
EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset); |
EXPECT_EQ(0u, rtp_header.extension.absoluteSendTime); |
+ EXPECT_FALSE(rtp_header.extension.voiceActivity); |
EXPECT_EQ(0u, rtp_header.extension.audioLevel); |
EXPECT_EQ(0u, rtp_header.extension.videoRotation); |
} |
@@ -504,9 +505,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) { |
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); |
+ EXPECT_TRUE(rtp_header.extension.voiceActivity); |
+ EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel); |
// Parse without map extension |
webrtc::RTPHeader rtp_header2; |
@@ -516,6 +516,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) { |
VerifyRTPHeaderCommon(rtp_header2); |
EXPECT_EQ(length, rtp_header2.headerLength); |
EXPECT_FALSE(rtp_header2.extension.hasAudioLevel); |
+ EXPECT_FALSE(rtp_header2.extension.voiceActivity); |
EXPECT_EQ(0u, rtp_header2.extension.audioLevel); |
} |
@@ -566,7 +567,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) { |
EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber); |
EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset); |
EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime); |
- EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel); |
+ EXPECT_TRUE(rtp_header.extension.voiceActivity); |
+ EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel); |
EXPECT_EQ(kTransportSequenceNumber, |
rtp_header.extension.transportSequenceNumber); |
@@ -584,6 +586,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) { |
EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset); |
EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime); |
+ EXPECT_FALSE(rtp_header2.extension.voiceActivity); |
EXPECT_EQ(0u, rtp_header2.extension.audioLevel); |
EXPECT_EQ(0u, rtp_header2.extension.transportSequenceNumber); |
} |