Move the VoE tests to use external transport instead of the built in udp transport

Review URL: https://webrtc-codereview.appspot.com/1223006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3708 4adac7df-926f-26a2-2b94-8c16560cd09d

tools/e2e_quality/audio/audio_e2e_harness.cc

@@ -15,17 +15,20 @@
 #include "gflags/gflags.h"
 #include "gtest/gtest.h"
 
-#include "voice_engine/include/voe_audio_processing.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
-#include "voice_engine/include/voe_base.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_audio_processing.h"
-#include "voice_engine/include/voe_hardware.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_hardware.h"
+#include "webrtc/voice_engine/include/voe_network.h"
 
 DEFINE_string(render, "render", "render device name");
 DEFINE_string(codec, "ISAC", "codec name");
 DEFINE_int32(rate, 16000, "codec sample rate in Hz");
 
 namespace webrtc {
-namespace {
+namespace test {
 
 void RunHarness() {
   VoiceEngine* voe = VoiceEngine::Create();
@@ -38,12 +41,18 @@ void RunHarness() {
   ASSERT_TRUE(codec != NULL);
   VoEHardware* hardware = VoEHardware::GetInterface(voe);
   ASSERT_TRUE(hardware != NULL);
+  VoENetwork* network = VoENetwork::GetInterface(voe);
+  ASSERT_TRUE(network != NULL);
 
   ASSERT_EQ(0, base->Init());
   int channel = base->CreateChannel();
   ASSERT_NE(-1, channel);
-  ASSERT_EQ(0, base->SetSendDestination(channel, 1234, "127.0.0.1"));
+
-  ASSERT_EQ(0, base->SetLocalReceiver(channel, 1234));
+  scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+      new VoiceChannelTransport(network, channel));
+
+  ASSERT_EQ(0, voice_channel_transport->SetSendDestination("127.0.0.1", 1234));
+  ASSERT_EQ(0, voice_channel_transport->SetLocalReceiver(1234));
 
   CodecInst codec_params = {0};
   bool codec_found = false;
@@ -90,10 +99,10 @@ void RunHarness() {
   }
 }
 
-}  // namespace
+}  // namespace test
 }  // namespace webrtc
 
 int main(int argc, char** argv) {
   google::ParseCommandLineFlags(&argc, &argv, true);
-  webrtc::RunHarness();
+  webrtc::test::RunHarness();
 }

tools/e2e_quality/e2e_quality.gyp

@@ -13,6 +13,7 @@
       'target_name': 'audio_e2e_harness',
       'type': 'executable',
       'dependencies': [
+        '<(webrtc_root)/test/channel_transport.gyp:channel_transport',
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',

webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h

@@ -13,9 +13,30 @@
 
 #include "before_initialization_fixture.h"
 #include "scoped_ptr.h"
+#include "webrtc/common_types.h"
 
 class TestErrorObserver;
 
+class LoopBackTransport : public webrtc::Transport {
+ public:
+  LoopBackTransport(webrtc::VoENetwork* voe_network)
+      : voe_network_(voe_network) {
+  }
+
+  virtual int SendPacket(int channel, const void *data, int len) {
+    voe_network_->ReceivedRTPPacket(channel, data, len);
+    return len;
+  }
+
+  virtual int SendRTCPPacket(int channel, const void *data, int len) {
+    voe_network_->ReceivedRTCPPacket(channel, data, len);
+    return len;
+  }
+
+ private:
+  webrtc::VoENetwork* voe_network_;
+};
+
 // This fixture initializes the voice engine in addition to the work
 // done by the before-initialization fixture. It also registers an error
 // observer which will fail tests on error callbacks. This fixture is

webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc

@@ -12,8 +12,6 @@
 
 #include <cstring>
 
-static const char* kLoopbackIp = "127.0.0.1";
-
 AfterStreamingFixture::AfterStreamingFixture()
     : channel_(voe_base_->CreateChannel()) {
   EXPECT_GE(channel_, 0);
@@ -30,7 +28,9 @@ AfterStreamingFixture::~AfterStreamingFixture() {
   voe_file_->StopPlayingFileAsMicrophone(channel_);
   PausePlaying();
 
+  EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(channel_));
   voe_base_->DeleteChannel(channel_);
+  delete transport_;
 }
 
 void AfterStreamingFixture::SwitchToManualMicrophone() {
@@ -59,8 +59,8 @@ void AfterStreamingFixture::ResumePlaying() {
 }
 
 void AfterStreamingFixture::SetUpLocalPlayback() {
-  EXPECT_EQ(0, voe_base_->SetSendDestination(channel_, 8000, kLoopbackIp));
+  transport_ = new LoopBackTransport(voe_network_);
-  EXPECT_EQ(0, voe_base_->SetLocalReceiver(0, 8000));
+  EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_, *transport_));
 
   webrtc::CodecInst codec;
   codec.channels = 1;

webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h

@@ -42,6 +42,8 @@ class AfterStreamingFixture : public AfterInitializationFixture {
 
  private:
   void SetUpLocalPlayback();
+
+  LoopBackTransport* transport_;
 };
 
 

webrtc/voice_engine/test/auto_test/standard/network_test.cc

@@ -23,79 +23,6 @@ class NetworkTest : public AfterStreamingFixture {
 
 using ::testing::Between;
 
-TEST_F(NetworkTest, GetSourceInfoReturnsPortsAndIpAfterReceivingPackets) {
-  // Give some time to send speech packets.
-  Sleep(200);
-
-  int rtp_port = 0;
-  int rtcp_port = 0;
-  char source_ip[32] = "127.0.0.1";
-
-  EXPECT_EQ(0, voe_network_->GetSourceInfo(channel_, rtp_port, rtcp_port,
-      source_ip));
-
-  EXPECT_EQ(kDefaultRtpPort, rtp_port);
-  EXPECT_EQ(kDefaultRtcpPort, rtcp_port);
-}
-
-TEST_F(NetworkTest, NoFilterIsEnabledByDefault) {
-  int filter_rtp_port = -1;
-  int filter_rtcp_port = -1;
-  char filter_ip[64] = { 0 };
-
-  EXPECT_EQ(0, voe_network_->GetSourceFilter(
-      channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
-
-  EXPECT_EQ(0, filter_rtp_port);
-  EXPECT_EQ(0, filter_rtcp_port);
-  EXPECT_STREQ("", filter_ip);
-}
-
-TEST_F(NetworkTest, ManualCanFilterRtpPort) {
-  TEST_LOG("No filter, should hear audio.\n");
-  Sleep(1000);
-
-  int port_to_block = kDefaultRtpPort + 10;
-  EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, port_to_block));
-
-  // Changes should take effect immediately.
-  int filter_rtp_port = -1;
-  int filter_rtcp_port = -1;
-  char filter_ip[64] = { 0 };
-
-  EXPECT_EQ(0, voe_network_->GetSourceFilter(
-      channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
-
-  EXPECT_EQ(port_to_block, filter_rtp_port);
-
-  TEST_LOG("Now filtering port %d, should not hear audio.\n", port_to_block);
-  Sleep(1000);
-
-  TEST_LOG("Removing filter, should hear audio.\n");
-  EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, 0));
-  Sleep(1000);
-}
-
-TEST_F(NetworkTest, ManualCanFilterIp) {
-  TEST_LOG("You should hear audio.\n");
-  Sleep(1000);
-
-  int rtcp_port_to_block = kDefaultRtcpPort + 10;
-  TEST_LOG("Filtering IP 10.10.10.10, should not hear audio.\n");
-  EXPECT_EQ(0, voe_network_->SetSourceFilter(
-      channel_, 0, rtcp_port_to_block, "10.10.10.10"));
-
-  int filter_rtp_port = -1;
-  int filter_rtcp_port = -1;
-  char filter_ip[64] = { 0 };
-  EXPECT_EQ(0, voe_network_->GetSourceFilter(
-      channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
-
-  EXPECT_EQ(0, filter_rtp_port);
-  EXPECT_EQ(rtcp_port_to_block, filter_rtcp_port);
-  EXPECT_STREQ("10.10.10.10", filter_ip);
-}
-
 TEST_F(NetworkTest,
     CallsObserverOnTimeoutAndRestartWhenPacketTimeoutNotificationIsEnabled) {
   // First, get rid of the default, asserting observer and install our observer.

webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc

@@ -101,10 +101,10 @@ class RtpRtcpTest : public AfterStreamingFixture {
     second_channel_ = voe_base_->CreateChannel();
     EXPECT_GE(second_channel_, 0);
 
-    EXPECT_EQ(0, voe_base_->SetSendDestination(
+    transport_ = new LoopBackTransport(voe_network_);
-        second_channel_, 8002, "127.0.0.1"));
+    EXPECT_EQ(0, voe_network_->RegisterExternalTransport(second_channel_,
-    EXPECT_EQ(0, voe_base_->SetLocalReceiver(
+                                                         *transport_));
-        second_channel_, 8002));
+
     EXPECT_EQ(0, voe_base_->StartReceive(second_channel_));
     EXPECT_EQ(0, voe_base_->StartPlayout(second_channel_));
     EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(second_channel_, 5678));
@@ -115,10 +115,13 @@ class RtpRtcpTest : public AfterStreamingFixture {
   }
 
   void TearDown() {
+    EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(second_channel_));
     voe_base_->DeleteChannel(second_channel_);
+    delete transport_;
   }
 
   int second_channel_;
+  LoopBackTransport* transport_;
 };
 
 void RtcpAppHandler::OnApplicationDataReceived(

webrtc/voice_engine/test/auto_test/voe_cpu_test.cc

@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "voe_cpu_test.h"
+
 #include <stdio.h>
 #include <string.h>
 #include <time.h>
@@ -16,9 +18,11 @@
 #include <conio.h>
 #endif
 
-#include "voe_cpu_test.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
 
 using namespace webrtc;
+using namespace test;
 
 namespace voetest {
 
@@ -45,6 +49,7 @@ int VoECpuTest::DoTest() {
   VoEFile* file = _mgr.FilePtr();
   VoECodec* codec = _mgr.CodecPtr();
   VoEAudioProcessing* apm = _mgr.APMPtr();
+  VoENetwork* voe_network = _mgr.NetworkPtr();
 
   int channel(-1);
   CodecInst isac;
@@ -59,8 +64,12 @@ int VoECpuTest::DoTest() {
   CHECK(base->Init());
   channel = base->CreateChannel();
 
-  CHECK(base->SetLocalReceiver(channel, 5566));
+  scoped_ptr<VoiceChannelTransport> voice_socket_transport(
-  CHECK(base->SetSendDestination(channel, 5566, "127.0.0.1"));
+      new VoiceChannelTransport(voe_network, channel));
+
+  CHECK(voice_socket_transport->SetSendDestination("127.0.0.1", 5566));
+  CHECK(voice_socket_transport->SetLocalReceiver(5566));
+
   CHECK(codec->SetRecPayloadType(channel, isac));
   CHECK(codec->SetSendCodec(channel, isac));
 
@@ -86,7 +95,6 @@ int VoECpuTest::DoTest() {
 
   base->DeleteChannel(channel);
   CHECK(base->Terminate());
-
   return 0;
 }
 

webrtc/voice_engine/test/auto_test/voe_extended_test.h

@@ -12,7 +12,14 @@
 #define WEBRTC_VOICE_ENGINE_VOE_EXTENDED_TEST_H
 
 #include "voe_standard_test.h"
-#include "modules/audio_device/include/audio_device.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/ref_count.h"
+#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
 
 namespace voetest {
 
@@ -425,35 +432,39 @@ class VoEExtendedTest : public VoiceEngineObserver,
   int TestRTP_RTCP();
   int TestVideoSync();
   int TestVolumeControl();
- public:
+
   int ErrorCode() const {
     return _errCode;
   }
   void ClearErrorCode() {
     _errCode = 0;
   }
+
  protected:
   // from VoiceEngineObserver
   void CallbackOnError(const int errCode, const int channel);
-  void CallbackOnTrace(const TraceLevel level, const char* message, const int length);
+  void CallbackOnTrace(const TraceLevel level, const char* message,
- protected:
+                       const int length);
+
   // from VoEConnectionObserver
   void OnPeriodicDeadOrAlive(const int channel, const bool alive);
+
  private:
-  void Play(int channel, unsigned int timeMillisec, bool addFileAsMicrophone = false,
+  void Play(int channel, unsigned int timeMillisec,
-            bool addTimeMarker = false);
+            bool addFileAsMicrophone = false, bool addTimeMarker = false);
   void Sleep(unsigned int timeMillisec, bool addMarker = false);
-  void StartMedia(int channel, int rtpPort, bool listen, bool playout, bool send);
+  void StartMedia(int channel, int rtpPort, bool listen, bool playout,
+                  bool send);
   void StopMedia(int channel);
   int RunMixingTest(int num_remote_channels, int num_local_channels,
                     int16_t input_value, int16_t max_output_value,
                     int16_t min_output_value);
- private:
+
   VoETestManager& _mgr;
- private:
   int _errCode;
   bool _alive;
   bool _listening[32];
+  scoped_ptr<webrtc::test::VoiceChannelTransport> voice_channel_transports_[32];
   bool _playing[32];
   bool _sending[32];
 };

webrtc/voice_engine/test/auto_test/voe_stress_test.cc

@@ -23,13 +23,16 @@
 #endif
 
 #include "webrtc/voice_engine/test/auto_test/voe_stress_test.h"
-#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
 
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"  // defines build macros
 
 using namespace webrtc;
+using namespace test;
 
 namespace voetest {
 
@@ -122,6 +125,7 @@ int VoEStressTest::StartStopTest() {
 
   // Get sub-API pointers
   VoEBase* base = _mgr.BasePtr();
+  VoENetwork* voe_network = _mgr.NetworkPtr();
 
   // Set trace
   //     VALIDATE_STRESS(base->SetTraceFileName(
@@ -147,9 +151,12 @@ int VoEStressTest::StartStopTest() {
   printf("Test will take approximately %d minutes. \n",
          numberOfLoops * loopSleep / 1000 / 60 + 1);
 
+  scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+      new VoiceChannelTransport(voe_network, 0));
+
   for (i = 0; i < numberOfLoops; ++i) {
-    VALIDATE_STRESS(base->SetLocalReceiver(0, 4800));
+    voice_channel_transport->SetSendDestination("127.0.0.1", 4800);
-    VALIDATE_STRESS(base->SetSendDestination(0, 4800, "127.0.0.1"));
+    voice_channel_transport->SetLocalReceiver(4800);
     VALIDATE_STRESS(base->StartReceive(0));
     VALIDATE_STRESS(base->StartPlayout(0));
     VALIDATE_STRESS(base->StartSend(0));
@@ -162,8 +169,9 @@ int VoEStressTest::StartStopTest() {
   }
   ANL();
 
-  VALIDATE_STRESS(base->SetLocalReceiver(0, 4800));
+  VALIDATE_STRESS(voice_channel_transport->SetSendDestination("127.0.0.1",
-  VALIDATE_STRESS(base->SetSendDestination(0, 4800, "127.0.0.1"));
+                                                              4800));
+  VALIDATE_STRESS(voice_channel_transport->SetLocalReceiver(4800));
   VALIDATE_STRESS(base->StartReceive(0));
   VALIDATE_STRESS(base->StartPlayout(0));
   VALIDATE_STRESS(base->StartSend(0));

webrtc/voice_engine/test/auto_test/voe_unit_test.cc

@@ -25,6 +25,7 @@
 #include "webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h"
 
 using namespace webrtc;
+using namespace test;
 
 namespace voetest {
 
@@ -248,13 +249,17 @@ int VoEUnitTest::StartMedia(int channel, int rtpPort, bool listen, bool playout,
                             bool send, bool fileAsMic, bool localFile) {
   VoEBase* base = _mgr.BasePtr();
   VoEFile* file = _mgr.FilePtr();
+  VoENetwork* voe_network = _mgr.NetworkPtr();
 
   _listening[channel] = false;
   _playing[channel] = false;
   _sending[channel] = false;
+  voice_channel_transports_[channel].reset(
+      new VoiceChannelTransport(voe_network, channel));
 
-  CHECK(base->SetLocalReceiver(channel, rtpPort));
+  CHECK(voice_channel_transports_[channel]->SetLocalReceiver(rtpPort));
-  CHECK(base->SetSendDestination(channel, rtpPort, "127.0.0.1"));
+  CHECK(voice_channel_transports_[channel]->SetSendDestination("127.0.0.1",
+                                                               rtpPort));
   if (listen) {
     _listening[channel] = true;
     CHECK(base->StartReceive(channel));

webrtc/voice_engine/test/auto_test/voe_unit_test.h

@@ -11,7 +11,9 @@
 #ifndef WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
 #define WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
 
-#include "voice_engine/test/auto_test/voe_standard_test.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
 
 namespace voetest {
 
@@ -57,6 +59,7 @@ class VoEUnitTest : public Encryption {
   bool _listening[32];
   bool _playing[32];
   bool _sending[32];
+  scoped_ptr<webrtc::test::VoiceChannelTransport> voice_channel_transports_[32];
 
  private:
   bool _extOnOff;

webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc

@@ -18,8 +18,6 @@
 #include <vector>
 
 #include "gtest/gtest.h"
-#include "test/testsupport/fileutils.h"
-
 #include "voe_errors.h"
 #include "voe_base.h"
 #include "voe_codec.h"
@@ -35,15 +33,17 @@
 #include "voe_network.h"
 #include "voe_neteq_stats.h"
 #include "engine_configurations.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/testsupport/fileutils.h"
 
 // Enable this this flag to run this test with hard coded
 // IP/Port/codec and start test automatically with key input
 // it could be useful in repeat tests.
 //#define DEBUG
 
-// #define EXTERNAL_TRANSPORT
-
 using namespace webrtc;
+using namespace test;
 
 #define VALIDATE                                                        \
   if (res != 0)                                                         \
@@ -70,29 +70,6 @@ VoENetEqStats* neteqst = NULL;
 
 void RunTest(std::string out_path);
 
-#ifdef EXTERNAL_TRANSPORT
-
-class my_transportation : public Transport
-{
-  int SendPacket(int channel,const void *data,int len);
-  int SendRTCPPacket(int channel, const void *data, int len);
-};
-
-int my_transportation::SendPacket(int channel,const void *data,int len)
-{
-  netw->ReceivedRTPPacket(channel, data, len);
-  return 0;
-}
-
-int my_transportation::SendRTCPPacket(int channel, const void *data, int len)
-{
-  netw->ReceivedRTCPPacket(channel, data, len);
-  return 0;
-}
-
-my_transportation my_transport;
-#endif
-
 class MyObserver : public VoiceEngineObserver {
  public:
    virtual void CallbackOnError(const int channel, const int err_code);
@@ -270,44 +247,24 @@ void RunTest(std::string out_path) {
   cnt++;
 
   int j = 0;
-#ifdef EXTERNAL_TRANSPORT
-  my_transportation ch0transport;
-  printf("Enabling external transport \n");
-  netw->RegisterExternalTransport(0, ch0transport);
-#else
   char ip[64];
 #ifdef DEBUG
   strcpy(ip, "127.0.0.1");
 #else
-  char localip[64];
-  netw->GetLocalIP(localip);
-  printf("local IP:%s\n", localip);
-
   printf("1. 127.0.0.1 \n");
   printf("2. Specify IP \n");
   ASSERT_EQ(1, scanf("%i", &i));
 
-  if (1 == i)
+  if (1 == i) {
     strcpy(ip, "127.0.0.1");
-  else {
+  } else {
     printf("Specify remote IP: ");
     ASSERT_EQ(1, scanf("%s", ip));
   }
 #endif
 
-  int colons(0);
+  int rPort = 8500;
-  while (ip[j] != '\0' && j < 64 && !(colons = (ip[j++] == ':')))
+#ifndef DEBUG
-    ;
-  if (colons) {
-    printf("Enabling IPv6\n");
-    res = netw->EnableIPv6(0);
-    VALIDATE;
-  }
-
-  int rPort;
-#ifdef DEBUG
-  rPort=8500;
-#else
   printf("Specify remote port (1=1234): ");
   ASSERT_EQ(1, scanf("%i", &rPort));
   if (1 == rPort)
@@ -315,23 +272,24 @@ void RunTest(std::string out_path) {
   printf("Set Send port \n");
 #endif
 
+  scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+      new VoiceChannelTransport(netw, chan));
+
   printf("Set Send IP \n");
-  res = base1->SetSendDestination(chan, rPort, ip);
+  res = voice_channel_transport->SetSendDestination(ip, rPort);
   VALIDATE;
 
-  int lPort;
+  int lPort = 8500;
-#ifdef DEBUG
+#ifndef DEBUG
-  lPort=8500;
-#else
   printf("Specify local port (1=1234): ");
   ASSERT_EQ(1, scanf("%i", &lPort));
   if (1 == lPort)
     lPort = 1234;
   printf("Set Rec Port \n");
 #endif
-  res = base1->SetLocalReceiver(chan, lPort);
+
+  res = voice_channel_transport->SetLocalReceiver(lPort);
   VALIDATE;
-#endif
 
   printf("\n");
   for (i = 0; i < codec->NumOfCodecs(); i++) {
@@ -367,12 +325,19 @@ void RunTest(std::string out_path) {
 #endif
   int channel_index = 0;
   std::vector<int> channels(kMaxNumChannels);
+  std::vector<scoped_ptr<VoiceChannelTransport> > voice_channel_transports;
+
   for (i = 0; i < kMaxNumChannels; ++i) {
     channels[i] = base1->CreateChannel();
     int port = rPort + (i + 1) * 2;
-    res = base1->SetSendDestination(channels[i], port, ip);
+
+    voice_channel_transports[i].reset(
+        new VoiceChannelTransport(netw, channels[i]));
+
+    printf("Set Send IP \n");
+    res = voice_channel_transports[i]->SetSendDestination(ip, port);
     VALIDATE;
-    res = base1->SetLocalReceiver(channels[i], port);
+    res = voice_channel_transports[i]->SetLocalReceiver(port);
     VALIDATE;
     res = codec->SetSendCodec(channels[i], cinst);
     VALIDATE;

webrtc/voice_engine/test/voice_engine_tests.gypi

@@ -20,7 +20,8 @@
         '<(DEPTH)/testing/gmock.gyp:gmock',
         '<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
         '<(webrtc_root)/test/libtest/libtest.gyp:libtest',
-      ],
+        '<(webrtc_root)/test/channel_transport.gyp:channel_transport',
+       ],
       'include_dirs': [
         'auto_test',
         'auto_test/fixtures',
@@ -102,6 +103,7 @@
         '<(DEPTH)/testing/gtest.gyp:gtest',
         'voice_engine_core',
         '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/test/channel_transport.gyp:channel_transport',
       ],
       'sources': [
         'cmd_test/voe_cmd_test.cc',