1/*
2 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11// Sets up a simple VoiceEngine loopback call with the default audio devices
12// and runs forever. Some parameters can be configured through command-line
13// flags.
14
15#include "gflags/gflags.h"
16#include "testing/gtest/include/gtest/gtest.h"
17
18#include "webrtc/system_wrappers/interface/scoped_ptr.h"
19#include "webrtc/test/channel_transport/include/channel_transport.h"
20#include "webrtc/voice_engine/include/voe_audio_processing.h"
21#include "webrtc/voice_engine/include/voe_base.h"
22#include "webrtc/voice_engine/include/voe_codec.h"
23#include "webrtc/voice_engine/include/voe_hardware.h"
24#include "webrtc/voice_engine/include/voe_network.h"
25
26DEFINE_string(render, "render", "render device name");
27DEFINE_string(codec, "ISAC", "codec name");
28DEFINE_int32(rate, 16000, "codec sample rate in Hz");
29
30namespace webrtc {
31namespace test {
32
33void RunHarness() {
34  VoiceEngine* voe = VoiceEngine::Create();
35  ASSERT_TRUE(voe != NULL);
36  VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe);
37  ASSERT_TRUE(audio != NULL);
38  VoEBase* base = VoEBase::GetInterface(voe);
39  ASSERT_TRUE(base != NULL);
40  VoECodec* codec = VoECodec::GetInterface(voe);
41  ASSERT_TRUE(codec != NULL);
42  VoEHardware* hardware = VoEHardware::GetInterface(voe);
43  ASSERT_TRUE(hardware != NULL);
44  VoENetwork* network = VoENetwork::GetInterface(voe);
45  ASSERT_TRUE(network != NULL);
46
47  ASSERT_EQ(0, base->Init());
48  int channel = base->CreateChannel();
49  ASSERT_NE(-1, channel);
50
51  scoped_ptr<VoiceChannelTransport> voice_channel_transport(
52      new VoiceChannelTransport(network, channel));
53
54  ASSERT_EQ(0, voice_channel_transport->SetSendDestination("127.0.0.1", 1234));
55  ASSERT_EQ(0, voice_channel_transport->SetLocalReceiver(1234));
56
57  CodecInst codec_params = {0};
58  bool codec_found = false;
59  for (int i = 0; i < codec->NumOfCodecs(); i++) {
60    ASSERT_EQ(0, codec->GetCodec(i, codec_params));
61    if (FLAGS_codec.compare(codec_params.plname) == 0 &&
62        FLAGS_rate == codec_params.plfreq) {
63      codec_found = true;
64      break;
65    }
66  }
67  ASSERT_TRUE(codec_found);
68  ASSERT_EQ(0, codec->SetSendCodec(channel, codec_params));
69
70  int num_devices = 0;
71  ASSERT_EQ(0, hardware->GetNumOfPlayoutDevices(num_devices));
72  char device_name[128] = {0};
73  char guid[128] = {0};
74  bool device_found = false;
75  int device_index;
76  for (device_index = 0; device_index < num_devices; device_index++) {
77    ASSERT_EQ(0, hardware->GetPlayoutDeviceName(device_index, device_name,
78                                                guid));
79    if (FLAGS_render.compare(device_name) == 0) {
80      device_found = true;
81      break;
82    }
83  }
84  ASSERT_TRUE(device_found);
85  ASSERT_EQ(0, hardware->SetPlayoutDevice(device_index));
86
87  // Disable all audio processing.
88  ASSERT_EQ(0, audio->SetAgcStatus(false));
89  ASSERT_EQ(0, audio->SetEcStatus(false));
90  ASSERT_EQ(0, audio->EnableHighPassFilter(false));
91  ASSERT_EQ(0, audio->SetNsStatus(false));
92
93  ASSERT_EQ(0, base->StartReceive(channel));
94  ASSERT_EQ(0, base->StartPlayout(channel));
95  ASSERT_EQ(0, base->StartSend(channel));
96
97  // Run forever...
98  while (1) {
99  }
100}
101
102}  // namespace test
103}  // namespace webrtc
104
105int main(int argc, char** argv) {
106  google::ParseCommandLineFlags(&argc, &argv, true);
107  webrtc::test::RunHarness();
108}
109