e30823911c
Review URL: https://webrtc-codereview.appspot.com/1223006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3708 4adac7df-926f-26a2-2b94-8c16560cd09d
109 lines
3.6 KiB
C++
109 lines
3.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// Sets up a simple VoiceEngine loopback call with the default audio devices
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// and runs forever. Some parameters can be configured through command-line
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// flags.
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#include "gflags/gflags.h"
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#include "gtest/gtest.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/test/channel_transport/include/channel_transport.h"
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#include "webrtc/voice_engine/include/voe_audio_processing.h"
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/voice_engine/include/voe_codec.h"
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#include "webrtc/voice_engine/include/voe_hardware.h"
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#include "webrtc/voice_engine/include/voe_network.h"
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DEFINE_string(render, "render", "render device name");
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DEFINE_string(codec, "ISAC", "codec name");
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DEFINE_int32(rate, 16000, "codec sample rate in Hz");
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namespace webrtc {
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namespace test {
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void RunHarness() {
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VoiceEngine* voe = VoiceEngine::Create();
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ASSERT_TRUE(voe != NULL);
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VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe);
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ASSERT_TRUE(audio != NULL);
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VoEBase* base = VoEBase::GetInterface(voe);
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ASSERT_TRUE(base != NULL);
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VoECodec* codec = VoECodec::GetInterface(voe);
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ASSERT_TRUE(codec != NULL);
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VoEHardware* hardware = VoEHardware::GetInterface(voe);
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ASSERT_TRUE(hardware != NULL);
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VoENetwork* network = VoENetwork::GetInterface(voe);
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ASSERT_TRUE(network != NULL);
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ASSERT_EQ(0, base->Init());
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int channel = base->CreateChannel();
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ASSERT_NE(-1, channel);
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scoped_ptr<VoiceChannelTransport> voice_channel_transport(
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new VoiceChannelTransport(network, channel));
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ASSERT_EQ(0, voice_channel_transport->SetSendDestination("127.0.0.1", 1234));
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ASSERT_EQ(0, voice_channel_transport->SetLocalReceiver(1234));
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CodecInst codec_params = {0};
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bool codec_found = false;
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for (int i = 0; i < codec->NumOfCodecs(); i++) {
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ASSERT_EQ(0, codec->GetCodec(i, codec_params));
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if (FLAGS_codec.compare(codec_params.plname) == 0 &&
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FLAGS_rate == codec_params.plfreq) {
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codec_found = true;
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break;
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}
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}
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ASSERT_TRUE(codec_found);
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ASSERT_EQ(0, codec->SetSendCodec(channel, codec_params));
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int num_devices = 0;
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ASSERT_EQ(0, hardware->GetNumOfPlayoutDevices(num_devices));
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char device_name[128] = {0};
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char guid[128] = {0};
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bool device_found = false;
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int device_index;
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for (device_index = 0; device_index < num_devices; device_index++) {
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ASSERT_EQ(0, hardware->GetPlayoutDeviceName(device_index, device_name,
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guid));
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if (FLAGS_render.compare(device_name) == 0) {
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device_found = true;
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break;
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}
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}
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ASSERT_TRUE(device_found);
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ASSERT_EQ(0, hardware->SetPlayoutDevice(device_index));
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// Disable all audio processing.
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ASSERT_EQ(0, audio->SetAgcStatus(false));
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ASSERT_EQ(0, audio->SetEcStatus(false));
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ASSERT_EQ(0, audio->EnableHighPassFilter(false));
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ASSERT_EQ(0, audio->SetNsStatus(false));
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ASSERT_EQ(0, base->StartReceive(channel));
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ASSERT_EQ(0, base->StartPlayout(channel));
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ASSERT_EQ(0, base->StartSend(channel));
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// Run forever...
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while (1) {
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}
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}
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} // namespace test
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} // namespace webrtc
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int main(int argc, char** argv) {
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google::ParseCommandLineFlags(&argc, &argv, true);
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webrtc::test::RunHarness();
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}
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