Separating voice activity flag from audio level in RtpHeaderExtension.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 302 matching lines @@
 
   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_FALSE(rtp_header.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header.extension.audioLevel);
   EXPECT_EQ(0u, rtp_header.extension.videoRotation);
 }
 
 TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
@@ skipping 164 matching lines @@
 
   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);
+  EXPECT_TRUE(rtp_header.extension.voiceActivity);
+  EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
   const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   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);
 }
 
 TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
   EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
   EXPECT_EQ(0,
             rtp_sender_->SetTransportSequenceNumber(kTransportSequenceNumber));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
@@ skipping 30 matching lines @@
   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_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);
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
   const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
 
   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_FALSE(rtp_header2.extension.hasTransportSequenceNumber);
 
   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);
 }
 
 TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
   EXPECT_CALL(mock_paced_sender_,
               SendPacket(PacedSender::kNormalPriority, _, kSeqNum, _, _, _)).
                   WillOnce(testing::Return(false));
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
@@ skipping 773 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc