Remove unused VoiceChannelTransport.

webrtc/voice_engine/test/channel_transport/channel_transport.cc

Index: webrtc/voice_engine/test/channel_transport/channel_transport.cc
diff --git a/webrtc/voice_engine/test/channel_transport/channel_transport.cc b/webrtc/voice_engine/test/channel_transport/channel_transport.cc
deleted file mode 100644
index 91c83d74a1c57805d52bc86733c2f8f7d515d953..0000000000000000000000000000000000000000
--- a/webrtc/voice_engine/test/channel_transport/channel_transport.cc
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- *  Copyright (c) 2013 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/voice_engine/test/channel_transport/channel_transport.h"
-
-#include <stdio.h>
-
-#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
-#include "webrtc/test/gtest.h"
-#endif
-#include "webrtc/voice_engine/test/channel_transport/udp_transport.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
-#undef NDEBUG
-#include <assert.h>
-#endif
-
-namespace webrtc {
-namespace test {
-
-VoiceChannelTransport::VoiceChannelTransport(VoENetwork* voe_network,
-                                             int channel)
-    : channel_(channel),
-      voe_network_(voe_network) {
-  uint8_t socket_threads = 1;
-  socket_transport_ = UdpTransport::Create(channel, socket_threads);
-  int registered = voe_network_->RegisterExternalTransport(channel,
-                                                           *socket_transport_);
-#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
-  EXPECT_EQ(0, registered);
-#else
-  assert(registered == 0);
-#endif
-}
-
-VoiceChannelTransport::~VoiceChannelTransport() {
-  voe_network_->DeRegisterExternalTransport(channel_);
-  UdpTransport::Destroy(socket_transport_);
-}
-
-void VoiceChannelTransport::IncomingRTPPacket(
-    const int8_t* incoming_rtp_packet,
-    const size_t packet_length,
-    const char* /*from_ip*/,
-    const uint16_t /*from_port*/) {
-  voe_network_->ReceivedRTPPacket(
-      channel_, incoming_rtp_packet, packet_length, PacketTime());
-}
-
-void VoiceChannelTransport::IncomingRTCPPacket(
-    const int8_t* incoming_rtcp_packet,
-    const size_t packet_length,
-    const char* /*from_ip*/,
-    const uint16_t /*from_port*/) {
-  voe_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet,
-                                   packet_length);
-}
-
-int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port) {
-  static const int kNumReceiveSocketBuffers = 500;
-  int return_value = socket_transport_->InitializeReceiveSockets(this,
-                                                                 rtp_port);
-  if (return_value == 0) {
-    return socket_transport_->StartReceiving(kNumReceiveSocketBuffers);
-  }
-  return return_value;
-}
-
-int VoiceChannelTransport::SetSendDestination(const char* ip_address,
-                                              uint16_t rtp_port) {
-  return socket_transport_->InitializeSendSockets(ip_address, rtp_port);
-}
-
-}  // namespace test
-}  // namespace webrtc