Removed vie_defines.h

webrtc/test/channel_transport/channel_transport.cc

Index: webrtc/test/channel_transport/channel_transport.cc
diff --git a/webrtc/test/channel_transport/channel_transport.cc b/webrtc/test/channel_transport/channel_transport.cc
index 25eb59d88708c6a7a94feb91b9ebd5dd7a85e40c..00ad7b2736835d1c1caa373e3697c4061e86b8c0 100644
--- a/webrtc/test/channel_transport/channel_transport.cc
+++ b/webrtc/test/channel_transport/channel_transport.cc
@@ -16,7 +16,6 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #endif
 #include "webrtc/test/channel_transport/udp_transport.h"
-#include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
@@ -66,10 +65,11 @@ void VoiceChannelTransport::IncomingRTCPPacket(
 }
 
 int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port) {
+  const int kNumReceiveSocketBuffers = 500;

[pbos-webrtc, 2015/10/29 15:39:40]

static

[mflodman, 2015/11/06 11:38:36]

Done.

   int return_value = socket_transport_->InitializeReceiveSockets(this,
                                                                  rtp_port);
   if (return_value == 0) {
-    return socket_transport_->StartReceiving(kViENumReceiveSocketBuffers);
+    return socket_transport_->StartReceiving(kNumReceiveSocketBuffers);
   }
   return return_value;
 }