Remove unused audio_e2e_harness.cc and infrastructure.

webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc

-/*
- *  Copyright (c) 2012 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.
- */
-
-// Sets up a simple VoiceEngine loopback call with the default audio devices
-// and runs forever. Some parameters can be configured through command-line
-// flags.
-
-#include <memory>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/test/channel_transport/channel_transport.h"
-#include "webrtc/voice_engine/include/voe_audio_processing.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 test {
-
-void RunHarness() {
-  VoiceEngine* voe = VoiceEngine::Create();
-  ASSERT_TRUE(voe != NULL);
-  VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe);
-  ASSERT_TRUE(audio != NULL);
-  VoEBase* base = VoEBase::GetInterface(voe);
-  ASSERT_TRUE(base != NULL);
-  VoECodec* codec = VoECodec::GetInterface(voe);
-  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);
-
-  std::unique_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;
-  for (int i = 0; i < codec->NumOfCodecs(); i++) {
-    ASSERT_EQ(0, codec->GetCodec(i, codec_params));
-    if (FLAGS_codec.compare(codec_params.plname) == 0 &&
-        FLAGS_rate == codec_params.plfreq) {
-      codec_found = true;
-      break;
-    }
-  }
-  ASSERT_TRUE(codec_found);
-  ASSERT_EQ(0, codec->SetSendCodec(channel, codec_params));
-
-  int num_devices = 0;
-  ASSERT_EQ(0, hardware->GetNumOfPlayoutDevices(num_devices));
-  char device_name[128] = {0};
-  char guid[128] = {0};
-  bool device_found = false;
-  int device_index;
-  for (device_index = 0; device_index < num_devices; device_index++) {
-    ASSERT_EQ(0, hardware->GetPlayoutDeviceName(device_index, device_name,
-                                                guid));
-    if (FLAGS_render.compare(device_name) == 0) {
-      device_found = true;
-      break;
-    }
-  }
-  ASSERT_TRUE(device_found);
-  ASSERT_EQ(0, hardware->SetPlayoutDevice(device_index));
-
-  // Disable all audio processing.
-  ASSERT_EQ(0, audio->SetAgcStatus(false));
-  ASSERT_EQ(0, audio->SetEcStatus(false));
-  ASSERT_EQ(0, audio->EnableHighPassFilter(false));
-  ASSERT_EQ(0, audio->SetNsStatus(false));
-
-  ASSERT_EQ(0, base->StartReceive(channel));
-  ASSERT_EQ(0, base->StartPlayout(channel));
-  ASSERT_EQ(0, base->StartSend(channel));
-
-  // Run forever...
-  while (1) {
-  }
-}
-
-}  // namespace test
-}  // namespace webrtc
-
-int main(int argc, char** argv) {
-  google::ParseCommandLineFlags(&argc, &argv, true);
-  webrtc::test::RunHarness();
-}