Remove Absolute Send Time from list of supported header extensions for audio streams.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 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 2032 matching lines @@
 }
 
 // Test support for audio level header extension.
 TEST_F(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) {
   TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri);
 }
 TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) {
   TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri);
 }
 
-// Test support for absolute send time header extension.
-TEST_F(WebRtcVoiceEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) {
-  TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri);
+// Test support for transport sequence number header extension.
+TEST_F(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) {
+  TestSetSendRtpHeaderExtensions(
+      webrtc::RtpExtension::kTransportSequenceNumberUri);
 }
-TEST_F(WebRtcVoiceEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) {
-  TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri);
+TEST_F(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) {
+  TestSetRecvRtpHeaderExtensions(
+      webrtc::RtpExtension::kTransportSequenceNumberUri);
 }
 
 // Test that we can create a channel and start sending on it.
 TEST_F(WebRtcVoiceEngineTestFake, Send) {
   EXPECT_TRUE(SetupSendStream());
   SetSendParameters(send_parameters_);
   SetSend(true);
   EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
   SetSend(false);
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
@@ skipping 1428 matching lines @@
       nullptr, webrtc::CreateBuiltinAudioDecoderFactory());
   webrtc::RtcEventLogNullImpl event_log;
   std::unique_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config(&event_log)));
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
                                            cricket::AudioOptions(), call.get());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.recv_codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));
 }