Configure VoE NACK through AudioSendStream::Config.

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 56 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_, 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()))
               .Times(1);
           EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
               .Times(1);
           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.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));
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
@@ skipping 61 matching lines @@
   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.cng_payload_type = 42;
   EXPECT_EQ(
       "{rtp: {ssrc: 1234, extensions: [{uri: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
-      "c_name: foo_name}, voe_channel_id: 1, cng_payload_type: 42}",
+      "nack: {rtp_history_ms: 0}, c_name: foo_name}, voe_channel_id: 1, "
+      "cng_payload_type: 42}",
       config.ToString());
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
                                         helper.congestion_controller());
 }
 
 TEST(AudioSendStreamTest, SendTelephoneEvent) {
@@ skipping 44 matching lines @@
       static_cast<internal::AudioState*>(helper.audio_state().get());
   VoiceEngineObserver* voe_observer =
       static_cast<VoiceEngineObserver*>(internal_audio_state);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
   EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 }
 }  // namespace test
 }  // namespace webrtc