Hooking up audio network adaptor to VoE.

webrtc/modules/audio_coding/test/PacketLossTest.cc

 /*
  *  Copyright (c) 2014 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 85 matching lines @@
 }
 
 bool SenderWithFEC::SetFEC(bool enable_fec) {
   if (_acm->SetCodecFEC(enable_fec) == 0) {
     return true;
   }
   return false;
 }
 
 bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
-  if (_acm->SetPacketLossRate(expected_loss_rate) == 0) {
-    expected_loss_rate_ = expected_loss_rate;
-    return true;
-  }
-  return false;
+  bool success = false;
+  _acm->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (!*encoder) {
+      // There is no existing encoder.
+      return;
+    }
+    (*encoder)->SetProjectedPacketLossRate(expected_loss_rate / 100.0f);
+    success = true;
+  });
+  return success;

[kwiberg-webrtc, 2016/10/06 09:46:13]

Excellent.

[minyue-webrtc, 2016/10/11 17:04:34]

Hi Karl,

I know that you liked this change, me too. but the codec creation in Sender uses
the old fashion and that prevents this to take place. I have to roll back.
Sorry.

 }
 
 PacketLossTest::PacketLossTest(int channels, int expected_loss_rate,
                                int actual_loss_rate, int burst_length)
     : channels_(channels),
       in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" :
                     "audio_coding/teststereo32kHz"),
       sample_rate_hz_(32000),
       sender_(new SenderWithFEC),
       receiver_(new ReceiverWithPacketLoss),
@@ skipping 38 matching lines @@
 
   receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
                    actual_loss_rate_, burst_length_);
   receiver_->Run();
   receiver_->Teardown();
   rtpFile.Close();
 #endif
 }
 
 }  // namespace webrtc