Hooking up audio network adaptor to VoE.

webrtc/modules/audio_coding/test/PacketLossTest.cc

Index: webrtc/modules/audio_coding/test/PacketLossTest.cc
diff --git a/webrtc/modules/audio_coding/test/PacketLossTest.cc b/webrtc/modules/audio_coding/test/PacketLossTest.cc
index 68f26150bced1c12e8957744459ae9dd36bbaa67..ae489552aefdea7b213e12f381031107a5a0973c 100644
--- a/webrtc/modules/audio_coding/test/PacketLossTest.cc
+++ b/webrtc/modules/audio_coding/test/PacketLossTest.cc
@@ -103,11 +103,16 @@ bool SenderWithFEC::SetFEC(bool enable_fec) {
 }
 
 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,