Clean up abs-send-time for audio.

webrtc/audio/audio_send_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 17 matching lines @@
 namespace test {
 namespace {
 
 using testing::_;
 using testing::Return;
 
 const int kChannelId = 1;
 const uint32_t kSsrc = 1234;
 const char* kCName = "foo_name";
 const int kAudioLevelId = 2;
-const int kAbsSendTimeId = 3;
 const int kTransportSequenceNumberId = 4;
 const int kEchoDelayMedian = 254;
 const int kEchoDelayStdDev = -3;
 const int kEchoReturnLoss = -65;
 const int kEchoReturnLossEnhancement = 101;
 const float kResidualEchoLikelihood = 0.0f;
 const unsigned int kSpeechInputLevel = 96;
 const CallStatistics kCallStats = {
     1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
@@ skipping 37 matching lines @@
         }));
 
     SetupMockForSetupSendCodec();
 
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 200;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
-    stream_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
     // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
     // calls from the default ctor behavior.
     stream_config_.send_codec_spec.codec_inst = kIsacCodec;
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
   CongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
   RtcEventLog* event_log() { return &event_log_; }
   MockVoiceEngine* voice_engine() { return &voice_engine_; }
 
   void SetupDefaultChannelProxy() {
     using testing::StrEq;
     channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
     EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
     EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
     EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
     EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
     EXPECT_CALL(*channel_proxy_,
-                SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
-        .Times(1);
-    EXPECT_CALL(*channel_proxy_,
                 SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
         .Times(1);
     EXPECT_CALL(*channel_proxy_,
                 EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
         .Times(1);
     EXPECT_CALL(*channel_proxy_,
                 RegisterSenderCongestionControlObjects(
                     congestion_controller_.pacer(),
                     congestion_controller_.GetTransportFeedbackObserver(),
                     congestion_controller_.packet_router()))
@@ skipping 75 matching lines @@
   BitrateAllocator bitrate_allocator_;
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   rtc::TaskQueue worker_queue_;
 };
 }  // namespace
 
 TEST(AudioSendStreamTest, ConfigToString) {
   AudioSendStream::Config config(nullptr);
   config.rtp.ssrc = kSsrc;
-  config.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
   config.rtp.c_name = kCName;
   config.voe_channel_id = kChannelId;
   config.min_bitrate_kbps = 12;
   config.max_bitrate_kbps = 34;
   config.send_codec_spec.nack_enabled = true;
   config.send_codec_spec.transport_cc_enabled = false;
   config.send_codec_spec.enable_codec_fec = true;
   config.send_codec_spec.enable_opus_dtx = false;
   config.send_codec_spec.opus_max_playback_rate = 32000;
   config.send_codec_spec.cng_payload_type = 42;
   config.send_codec_spec.cng_plfreq = 56;
   config.send_codec_spec.min_ptime_ms = 20;
   config.send_codec_spec.max_ptime_ms = 60;
   config.send_codec_spec.codec_inst = kIsacCodec;
+  config.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
   EXPECT_EQ(
       "{rtp: {ssrc: 1234, extensions: [{uri: "
-      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
-      "nack: {rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
+      "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: "
+      "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
       "voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, "
       "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
       "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "
       "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: "
       "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: "
       "320, channels: 1, rate: 32000}}}",
       config.ToString());
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
@@ skipping 130 matching lines @@
   EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _))
       .WillOnce(Return(0));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
 }
 
 }  // namespace test
 }  // namespace webrtc