Clean up abs-send-time for audio.

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 33 matching lines @@
   audio_decode_stats.decoded_plc_cng = 789;
   audio_decode_stats.decoded_muted_output = 987;
   return audio_decode_stats;
 }
 
 const int kChannelId = 2;
 const uint32_t kRemoteSsrc = 1234;
 const uint32_t kLocalSsrc = 5678;
 const size_t kOneByteExtensionHeaderLength = 4;
 const size_t kOneByteExtensionLength = 4;
-const int kAbsSendTimeId = 2;
 const int kAudioLevelId = 3;
 const int kTransportSequenceNumberId = 4;
 const int kJitterBufferDelay = -7;
 const int kPlayoutBufferDelay = 302;
 const unsigned int kSpeechOutputLevel = 99;
 const CallStatistics kCallStats = {
     345,  678,  901, 234, -12, 3456, 7890, 567, 890, 123};
 const CodecInst kCodecInst = {
     123, "codec_name_recv", 96000, -187, 0, -103};
 const NetworkStatistics kNetworkStats = {
@@ skipping 18 matching lines @@
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);
 
     EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
         .WillOnce(Invoke([this](int channel_id) {
           EXPECT_FALSE(channel_proxy_);
           channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
           EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
           EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
           EXPECT_CALL(*channel_proxy_,
-              SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
-                  .Times(1);
-          EXPECT_CALL(*channel_proxy_,
               SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_,
               EnableReceiveTransportSequenceNumber(kTransportSequenceNumberId))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_,
               RegisterReceiverCongestionControlObjects(&packet_router_))
                   .Times(1);
           EXPECT_CALL(congestion_controller_, packet_router())
               .WillOnce(Return(&packet_router_));
@@ skipping 12 matching lines @@
               .Times(1)
               .After(expect_set);
           return channel_proxy_;
         }));
     EXPECT_CALL(voice_engine_, StopPlayout(kChannelId)).WillOnce(Return(0));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 300;
     stream_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));

[the sun, 2016/10/27 16:09:17]

Add audio level instead, and keep the test of ToString().

[stefan-webrtc, 2016/10/31 15:40:27]

Audio level is already there. This is unrelated to the ToString() test.

Adding something to ToString test below anyway.

[the sun, 2016/11/01 08:39:40]

Thank you!

-    stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
     stream_config_.decoder_factory = decoder_factory_;
   }
 
   MockCongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
@@ skipping 97 matching lines @@
   ByteWriter<uint16_t>::WriteBigEndian(&packet[2], 6);
   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::kAbsSendTimeUri, kAbsSendTimeId));
   config.voe_channel_id = kChannelId;
   EXPECT_EQ(
-      "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, "
-      "nack: {rtp_history_ms: 0}, extensions: [{uri: "
-      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
-      "rtcp_send_transport: nullptr, "
+      "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
+      "{rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: nullptr, "
       "voe_channel_id: 2}",
       config.ToString());
 }
 
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state(),
       helper.event_log());
 }
 
 MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
-  return arg.extension.hasAbsoluteSendTime ==
-             expected_extension.hasAbsoluteSendTime &&
-         arg.extension.absoluteSendTime ==
-             expected_extension.absoluteSendTime &&
-         arg.extension.hasTransportSequenceNumber ==
+  return arg.extension.hasTransportSequenceNumber ==
              expected_extension.hasTransportSequenceNumber &&
          arg.extension.transportSequenceNumber ==
              expected_extension.transportSequenceNumber;
 }
 
 TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
   ConfigHelper helper;
   helper.config().rtp.transport_cc = true;
   helper.SetupMockForBweFeedback(true);
   internal::AudioReceiveStream recv_stream(
@@ skipping 85 matching lines @@
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state(),
       helper.event_log());
   EXPECT_CALL(*helper.channel_proxy(),
       SetChannelOutputVolumeScaling(FloatEq(0.765f)));
   recv_stream.SetGain(0.765f);
 }
 }  // namespace test
 }  // namespace webrtc