WebRtc中VoiceEngine可以完成大部分的VOIP相关人物，包括采集、自动增益、噪声消除、回声抑制、编解码、ＲＴＰ传输。下边我们通过代码来解析Voe中处理流程；
创建VoiceEngine和VoEBase
[cpp]
VoiceEngine* _vePtr = VoiceEngine::Create(); //创建VoiceEngine
VoEBase* _veBasePtr = VoEBase::GetInterface(_vePtr); //创建VoeBase 所有Voe相关操作通过这个共有类
_veBasePtr->Init() //创建整个Voe处理线程
VoiceEngine* _vePtr = VoiceEngine::Create(); //创建VoiceEngine
VoEBase* _veBasePtr = VoEBase::GetInterface(_vePtr); //创建VoeBase 所有Voe相关操作通过这个共有类
_veBasePtr->Init() //创建整个Voe处理线程
重点就在_veBasePtr->Init() 它会创建voe线程，线程负责采集、数字信号处理、编码、rtp传输。
[cpp]
int VoEBaseImpl::Init(AudioDeviceModule* external_adm, AudioProcessing* audioproc)
{
_shared->process_thread()； //创建voe线程
_shared->process_thread()->Start()；
_shared->audio_device()->Init()；
}
int VoEBaseImpl::Init(AudioDeviceModule* external_adm, AudioProcessing* audioproc)
{
_shared->process_thread()； //创建voe线程
_shared->process_thread()->Start()；
_shared->audio_device()->Init()；
}audio_device()->Init()重载了int32_t AudioDeviceWindowsWave::Init()（windowns平台），别的平台是别的函数，基本差不多，在这个Init中，创建了ThreadProcess线程，ThreadProcess线程负责所有的音频流程，从设备获取音频数据包。
[cpp]
bool AudioDeviceWindowsWave::ThreadProcess()
{
while ((nRecordedBytes = RecProc(recTime)) > 0)；
}
bool AudioDeviceWindowsWave::ThreadProcess()
{
while ((nRecordedBytes = RecProc(recTime)) > 0)；
}处理过程在RecProc
[cpp]
int32_t AudioDeviceWindowsWave::RecProc(LONGLONG& consumedTime)
{
_ptrAudioBuffer->DeliverRecordedData(); <SPAN style="FONT-FAMILY: Arial, Helvetica, sans-serif">}</SPAN>
int32_t AudioDeviceWindowsWave::RecProc(LONGLONG& consumedTime)
{
_ptrAudioBuffer->DeliverRecordedData(); }
[cpp]
int32_t AudioDeviceBuffer::DeliverRecordedData()
{
_ptrCbAudioTransport->RecordedDataIsAvailable();
}
int32_t AudioDeviceBuffer::DeliverRecordedData()
{
_ptrCbAudioTransport->RecordedDataIsAvailable();
}
RecordedDataIsAvailable是虚函数，被VoeBase重载
[cpp]
int32_t VoEBaseImpl::RecordedDataIsAvailable(
const void* audioSamples,
uint32_t nSamples,
uint8_t nBytesPerSample,
uint8_t nChannels,
uint32_t samplesPerSec,
uint32_t totalDelayMS,
int32_t clockDrift,
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel)
{
_shared->transmit_mixer()->DemuxAndMix();
_shared->transmit_mixer()->EncodeAndSend();
}
int32_t VoEBaseImpl::RecordedDataIsAvailable(
const void* audioSamples,
uint32_t nSamples,
uint8_t nBytesPerSample,
uint8_t nChannels,
uint32_t samplesPerSec,
uint32_t totalDelayMS,
int32_t clockDrift,
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel)
{
_shared->transmit_mixer()->DemuxAndMix();
_shared->transmit_mixer()->EncodeAndSend();
}
DemuxAndMix() 从字面意思是分路与混合，这个函数，主要负责AudioProcess的所有过程，包括Aec,Aecm,AGC,DTMF，遍历所有channel;
[cpp]
TransmitMixer::DemuxAndMix()
{
Channel* channelPtr = sc.GetFirstChannel(iterator);
while (channelPtr != NULL)
{
if (channelPtr->InputIsOnHold())
{
channelPtr->UpdateLocalTimeStamp();
} else if (channelPtr->Sending())
{
// Demultiplex makes a copy of its input.
channelPtr->Demultiplex(_audioFrame);
channelPtr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_);
}
channelPtr = sc.GetNextChannel(iterator);
}
}
TransmitMixer::DemuxAndMix()
{
Channel* channelPtr = sc.GetFirstChannel(iterator);
while (channelPtr != NULL)
{
if (channelPtr->InputIsOnHold())
{
channelPtr->UpdateLocalTimeStamp();
} else if (channelPtr->Sending())
{
// Demultiplex makes a copy of its input.
channelPtr->Demultiplex(_audioFrame);
channelPtr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_);
}
channelPtr = sc.GetNextChannel(iterator);
}
}
Channel::Demutiplex(),基本上没有什么具体任务，就是把audioFrame里边的数据拷贝到channel自身, webrtc是client解决方案，对于client只认为有一个audio source,但可以有多个channel，每个channel中都有audio process，所以需要把数据copy到每个channel.
只有就是数据处理 PrepareEncodeAndSend()
[cpp]
Channel::PrepareEncodeAndSend(int mixingFrequency)
{
if (_inputFilePlaying)
{
MixOrReplaceAudioWithFile(mixingFrequency); //如果使用了voeFile::PlayFileAsMic()；则从文件读取10ms数据，并覆盖audio buffer
}
if (_mute)
{
AudioFrameOperations::Mute(_audioFrame);//当然如果设置mutex，则memset 0
}
if (_inputExternalMedia)
{
_inputExternalMediaCallbackPtr->Process(); //所过设置了ExternalMedia,自己的audio处理过程，就是在这里调用的
}
InsertInbandDtmfTone(); //添加DTMF音频
_rtpAudioProc->ProcessStream(&_audioFrame); // 真正的GIPS牛逼代码，audio process过程: Aec Aecm AGC
}
Channel::PrepareEncodeAndSend(int mixingFrequency)
{
if (_inputFilePlaying)
{
MixOrReplaceAudioWithFile(mixingFrequency); //如果使用了voeFile::PlayFileAsMic()；则从文件读取10ms数据，并覆盖audio buffer
}
if (_mute)
{
AudioFrameOperations::Mute(_audioFrame);//当然如果设置mutex，则memset 0
}
if (_inputExternalMedia)
{
_inputExternalMediaCallbackPtr->Process(); //所过设置了ExternalMedia,自己的audio处理过程，就是在这里调用的
}
InsertInbandDtmfTone(); //添加DTMF音频
_rtpAudioProc->ProcessStream(&_audioFrame); // 真正的GIPS牛逼代码，audio process过程: Aec Aecm AGC
}
int AudioProcessingImpl::ProcessStream(AudioFrame* frame) 就是上述调用的_rtpAudioProc->ProcessStream();
以上是DemuxAndMix（）过程，之后就是EncodeAndSend()过程，至此整个voe数据处理流程分析结束；
关于Audio Process则是另外一个大话题；
总结一下几点：
1. VoeBase提供大部分的对外接口
2. Channel：继承了大部分的音频功能；
