Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/

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
new file mode 100644
index 0000000000000000000000000000000000000000..38eefe54a2b8b8a3c9a9af50ed6f9cb7cd694388
--- /dev/null
+++ b/webrtc/test/channel_transport/channel_transport.cc
@@ -0,0 +1,83 @@
+/*
+ *  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/test/channel_transport/channel_transport.h"
+
+#include <stdio.h>
+
+#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
+#include "testing/gtest/include/gtest/gtest.h"
+#endif
+#include "webrtc/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