Move some receive stream configuration into webrtc::AudioReceiveStream.

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 20 matching lines @@
   audio_decode_stats.decoded_plc = 123;
   audio_decode_stats.decoded_cng = 456;
   audio_decode_stats.decoded_plc_cng = 789;
   return audio_decode_stats;
 }
 
 const int kChannelId = 2;
 const uint32_t kRemoteSsrc = 1234;
 const uint32_t kLocalSsrc = 5678;
 const size_t kAbsoluteSendTimeLength = 4;
-const int kAbsSendTimeId = 3;
+const int kAbsSendTimeId = 2;
+const int kAudioLevelId = 3;
 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, -198, -103};
 const NetworkStatistics kNetworkStats = {
     123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
 
 struct ConfigHelper {
   ConfigHelper() {
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);
+
+    EXPECT_CALL(voice_engine_, SetLocalSSRC(kChannelId, kLocalSsrc))
+        .WillOnce(Return(0));
+    EXPECT_CALL(voice_engine_,
+        SetReceiveAbsoluteSenderTimeStatus(kChannelId, true, kAbsSendTimeId))
+            .WillOnce(Return(0));
+    EXPECT_CALL(voice_engine_,
+        SetReceiveAudioLevelIndicationStatus(kChannelId, true, kAudioLevelId))
+            .WillOnce(Return(0));
     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));
+    stream_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+}
 
   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
     return &remote_bitrate_estimator_;
   }
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }
 
   void SetupMockForGetStats() {
     using testing::DoAll;
@@ skipping 61 matching lines @@
 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));
   config.voe_channel_id = kChannelId;
   config.combined_audio_video_bwe = true;
   EXPECT_EQ(
       "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
-      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, "
+      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
       "receive_transport: nullptr, rtcp_send_transport: nullptr, "
       "voe_channel_id: 2, combined_audio_video_bwe: true}",
       config.ToString());
 }
 
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
 }
 
 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
   ConfigHelper helper;
   helper.config().combined_audio_video_bwe = true;
-  helper.config().rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
   internal::AudioReceiveStream recv_stream(
       helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
   uint8_t rtp_packet[30];
   const int kAbsSendTimeValue = 1234;
   CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
   PacketTime packet_time(5678000, 0);
   const size_t kExpectedHeaderLength = 20;
   EXPECT_CALL(*helper.remote_bitrate_estimator(),
               IncomingPacket(packet_time.timestamp / 1000,
                              sizeof(rtp_packet) - kExpectedHeaderLength,
@@ skipping 39 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