Remove use of RtpHeaderExtension and clean up

webrtc/audio/audio_receive_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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 90 matching lines @@
           EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
               .Times(1);
           EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
               .Times(1);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+        RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+        RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
-        RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
+        RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
   }
 
   MockCongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
     return &remote_bitrate_estimator_;
   }
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
@@ skipping 91 matching lines @@
   ByteWriter<uint32_t>::WriteBigEndian(&packet[4], kLocalSsrc);
   return packet;
 }
 }  // namespace
 
 TEST(AudioReceiveStreamTest, ConfigToString) {
   AudioReceiveStream::Config config;
   config.rtp.remote_ssrc = kRemoteSsrc;
   config.rtp.local_ssrc = kLocalSsrc;
   config.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+      RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
   config.voe_channel_id = kChannelId;
   EXPECT_EQ(
-      "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
+      "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{uri: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}], "
       "transport_cc: off}, "
       "rtcp_send_transport: nullptr, "
       "voe_channel_id: 2}",
       config.ToString());
 }
 
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
@@ skipping 90 matching lines @@
   EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
   EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
   EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
   EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
             stats.capture_start_ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc