blob: 4d443773fd5fbd3cde548f47832ddc067d458c18 [file] [log] [blame]
/*
* 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 "gflags/gflags.h"
#include "gtest/gtest.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/channel_transport/include/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);
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;
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();
}