Regression test for issue where Opus DTX status was being forgotten.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 1530 matching lines @@
   int ret = enable_dtx ? audio_coding_->EnableOpusDtx()
                        : audio_coding_->DisableOpusDtx();
   if (ret != 0) {
     _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR,
                                        kTraceError, "SetOpusDtx() failed");
     return -1;
   }
   return 0;
 }
 
+int Channel::GetOpusDtx(bool* enabled) {
+  int success = -1;
+  audio_coding_->QueryEncoder([&](AudioEncoder const* encoder) {
+    if (encoder) {
+      *enabled = encoder->GetDtx();
+      success = 0;
+    }
+  });
+  return success;
+}
+
 int32_t Channel::RegisterExternalTransport(Transport* transport) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterExternalTransport()");
 
   rtc::CritScope cs(&_callbackCritSect);
   if (_externalTransport) {
     _engineStatisticsPtr->SetLastError(
         VE_INVALID_OPERATION, kTraceError,
         "RegisterExternalTransport() external transport already enabled");
     return -1;
@@ skipping 2001 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc