WebRtc 音频引擎-linux demo

Google收购了著名的音频技术公司GIPS后，基于其强大的音频技术，实现了WebRtc的Voice Engine，即语音处理引擎。本文主要介绍WebRTC 中Voice Engine中音频技术相关的实现，并结合具体实例，介绍如何利用voice engine实现自己的VoIP音频处理引擎。

本文主要介绍如何在linux下搭建一个可以自己调试的基于WebRTC的voiceEngine。

1.VoiceEngine Demo 目录树

下面是一个小的VoiceEngine目录树：

.
├── include
│   ├── channel_transport.h
│   ├── common_types.h
│   ├── typedefs.h
│   ├── udp_transport.h
│   ├── voe_audio_processing.h
│   ├── voe_base.h
│   ├── voe_call_report.h
│   ├── voe_codec.h
│   ├── voe_dtmf.h
│   ├── voe_encryption.h
│   ├── voe_errors.h
│   ├── voe_external_media.h
│   ├── voe_file.h
│   ├── voe_hardware.h
│   ├── voe_neteq_stats.h
│   ├── voe_network.h
│   ├── voe_rtp_rtcp.h
│   ├── voe_video_sync.h
│   └── voe_volume_control.h
├── lib
│   ├── libaudio_coding_module.a
│   ├── libaudio_conference_mixer.a
│   ├── libaudio_device.a
│   ├── libaudioproc_debug_proto.a
│   ├── libaudio_processing.a
│   ├── libaudio_processing_sse2.a
│   ├── libchannel_transport.a
│   ├── libCNG.a
│   ├── libcommon_video.a
│   ├── libG711.a
│   ├── libG722.a
│   ├── libgtest.a
│   ├── libgtest_main.a
│   ├── libiLBC.a
│   ├── libiSAC.a
│   ├── libiSACFix.a
│   ├── libmedia_file.a
│   ├── libNetEq.a
│   ├── libopus.a
│   ├── libpaced_sender.a
│   ├── libPCM16B.a
│   ├── libprotobuf_lite.a
│   ├── libresampler.a
│   ├── librtp_rtcp.a
│   ├── libsignal_processing.a
│   ├── libsystem_wrappers.a
│   ├── libvad.a
│   ├── libvoice_engine_core.a
│   ├── libwebrtc_opus.a
│   └── libwebrtc_utility.a
├── Makefile
├── out
│   └── Debug
│       ├── client_recv
│       └── client_send
└── src
    ├── client_recv.cpp
    └── client_send.cpp

 

其中，src目录下的client_send和client_recv是基于WebRTC VoiceEngine实现的两个Demo，一个发送音频数据、一个接收音频数据。

2.工程Makefile

下面是Voiceengine工程编译的Makefile文件

#WebRTC VoiceEngine Test => Makefile                                                                                                  

CC = g++ 
CFLAGS= -Wall -g
VPATH = src:include
lib= -L lib 

obj=out/Debug/client_send  out/Debug/client_recv

depens= -lvoice_engine_core -laudio_device -lresampler \
        -laudio_conference_mixer\
        -laudio_processing  \
        -laudio_coding_module -lrtp_rtcp\
        -lNetEq -lCNG -lG722 -liLBC \
        -lG711 -liSAC -lPCM16B \
        -lsignal_processing \
        -lvad -laudioproc_debug_proto\
        -lprotobuf_lite -laudio_processing_sse2\
        -lwebrtc_opus -lopus  -lpaced_sender\
        -liSACFix -lmedia_file \
        -lwebrtc_utility -lchannel_transport -lgtest\
        -lpthread -lsystem_wrappers -lrt -ldl\

all:${obj}

out/Debug/client_send:client_send.cpp
        ${CC} ${CFLAGS} -o $@ $< -Iinclude  ${lib} ${depens}
        
out/Debug/client_recv:client_recv.cpp 
        ${CC} ${CFLAGS} -o $@ $< -Iinclude  ${lib} ${depens}

.PHONY:clean
clean:
        rm -rf *.o ${obj}

 

其中，静态库的链接顺序不能随便修改，由于静态库之间存在依赖关系。具体原因可以看这里

3.client_recv Demo

/*
*  WebRTC VoiceEngine Test => client_recv
*  
*  @date:13.06.2013
*  @author:hongliang
*  @mail:lhl_nciae@sina.cn
*/

#include<iostream>
#include"voe_base.h"
#include"voe_network.h"
#include"voe_hardware.h"
#include"voe_errors.h""
#include"channel_transport.h"


using namespace webrtc;

int main(int argc , char *argv[])
{
	//Create VoiceEngine
	VoiceEngine* voe = VoiceEngine::Create();

	//Init base
	VoEBase* base = VoEBase::GetInterface(voe);
	base->Init();

	//hardware
	VoEHardware* hardware = VoEHardware::GetInterface(voe);

	int nRec = 0;
	char devName[128] = {0};
	char guidName[128] = {0};
	int ret = 0;

	ret = hardware->GetNumOfRecordingDevices(nRec);

	if(ret != 0)
	{
		std::cout << "GetNumOfRecordingDevice error:" << base->LastError() << std::endl;
	}

	for (int idx = 0; idx < nRec; idx++)
	{
		hardware->GetRecordingDeviceName(idx , devName , guidName);
		std::cout << "GetRecordingDeviceName=> " << "name:" << devName << " guidname:" << guidName <<std::endl;
	}

	//Create Channel
	int ch = base->CreateChannel();
	if(ch != -1)
	{
		std::cout << "Create channel #" << ch << std::endl;
	}
	
	//Create Voice Channel transport
	VoENetwork* voe_network = VoENetwork::GetInterface(voe);
	
	test::VoiceChannelTransport voe_vct = test::VoiceChannelTransport(voe_network , ch);

	//recv
	voe_vct.SetLocalReceiver(12345);
	base->StartReceive(ch);
	base->StartPlayout(ch);

	std::cout << "Start Receice from channel:" << ch << std::endl;

	while(1)
	{
	}	
	

	//Release resource
	base->DeleteChannel(ch);
	base->Terminate();
	base->Release();
	hardware->Release();
	VoiceEngine::Delete(voe);

	return 0;
}

 

4.client_send Demo

#include<iostream>
#include"voe_base.h"
#include"voe_network.h"
#include"voe_hardware.h"
#include"voe_errors.h"
#include"voe_rtp_rtcp.h"
#include"channel_transport.h"

using namespace webrtc;

int main(int argc ,char * argv[])
{
	int ret;
	//Create VoiceEngine
	VoiceEngine *voe = VoiceEngine::Create();

	//Init base
	VoEBase* base = VoEBase::GetInterface(voe);
	base->Init();

	//handware
	int nRec = 0;
	char devName[128] = {0};
	char guidName[128] = {0};
	
	VoEHardware* hardware = VoEHardware::GetInterface(voe);
	hardware->GetNumOfRecordingDevices(nRec);
	std::cout << "Get num of recordingdevice:" << nRec << std::endl;	
	for(int idx = 0; idx < nRec; idx++)
	{
		hardware->GetRecordingDeviceName(idx , devName , guidName);
		std::cout << "GetRecordingName(" << idx << ")  " << "name:" << devName << "  guidName:" << guidName << std::endl;
	}

	//Create Channel
	int ch = base->CreateChannel();
	if(ch == -1)
	{
		std::cout << "create channel error:" << base->LastError() << std::endl;
		return -1;
	}	

	std::cout << "create channel#" << ch << std::endl;
	//Create Voice Channel transport
	VoENetwork* voe_network = VoENetwork::GetInterface(voe);
	
	test::VoiceChannelTransport voe_ctp = test::VoiceChannelTransport(voe_network , ch);

	//send
	voe_ctp.SetSendDestination("192.168.1.1" , 12345);
//	base->SetSendDestination(ch , "192.168.1.1" , 12345);

	ret = base->StartSend(ch);	
	if(ret == -1)
	{
		std::cout << "Start send error:" << base->LastError() << std::endl;
		return -1;
	}

	std::cout << "Start send on channel#" << ch << std::endl;

	//Release Resource
	base->DeleteChannel(ch);
	base->Terminate();
	hardware->Release();
	VoiceEngine::Delete(voe);

	return 0;
}