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:[email protected] */ #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;
}