Adding audio DTX to video loopback test.

webrtc/video/video_quality_test.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.
  */
 #include "webrtc/video/video_quality_test.h"
@@ skipping 1016 matching lines @@
   std::set<VideoEncoder*> live_encoders_;
 };
 
 VideoQualityTest::VideoQualityTest()
     : clock_(Clock::GetRealTimeClock()), receive_logs_(0), send_logs_(0) {}
 
 VideoQualityTest::Params::Params()
     : call({false, Call::Config::BitrateConfig()}),
       video({false, 640, 480, 30, 50, 800, 800, false, "VP8", 1, -1, 0, false,
              false, "", ""}),
-      audio({false, false}),
+      audio({false, false, false}),
       screenshare({false, 10, 0}),
       analyzer({"", 0.0, 0.0, 0, "", ""}),
       pipe(),
       logs(false),
       ss({std::vector<VideoStream>(), 0, 0, -1, std::vector<SpatialLayer>()}),
       num_thumbnails(0) {}
 
 VideoQualityTest::Params::~Params() = default;
 
 void VideoQualityTest::TestBody() {}
@@ skipping 655 matching lines @@
   audio_send_config_.rtp.extensions.clear();
   if (params_.call.send_side_bwe) {
     audio_send_config_.rtp.extensions.push_back(
         webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
                              test::kTransportSequenceNumberExtensionId));
     audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps;
     audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps;
   }
   audio_send_config_.send_codec_spec.codec_inst =
       CodecInst{120, "OPUS", 48000, 960, 2, 64000};
-
+  audio_send_config_.send_codec_spec.enable_opus_dtx = params_.audio.dtx;
   audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
 
   AudioReceiveStream::Config audio_config;
   audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
   audio_config.rtcp_send_transport = transport;
   audio_config.voe_channel_id = receive_channel_id;
   audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
   audio_config.rtp.transport_cc = params_.call.send_side_bwe;
   audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
   audio_config.decoder_factory = decoder_factory_;
@@ skipping 157 matching lines @@
   if (!params_.video.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 }  // namespace webrtc