下面将以实现一个音频通话功能为示例详细介绍VoiceEngine的使用，在文末将附上相应源码的下载地址。这里参考的是voiceengine\voe_cmd_test。

第一步是创建VoiceEngine和相关的sub-apis

// Create VoiceEngine related instance

webrtc::VoiceEngine* ptrVoE = NULL;

ptrVoE = webrtc::VoiceEngine::Create();

webrtc::VoEBase* ptrVoEBase = NULL;

ptrVoEBase = webrtc::VoEBase::GetInterface(ptrVoE);

webrtc::VoECodec* ptrVoECodec = NULL;

ptrVoECodec = webrtc::VoECodec::GetInterface(ptrVoE);

webrtc::VoEAudioProcessing* ptrVoEAp = NULL;

ptrVoEAp = webrtc::VoEAudioProcessing::GetInterface(ptrVoE);

webrtc::VoEVolumeControl* ptrVoEVolume = NULL;

ptrVoEVolume = webrtc::VoEVolumeControl::GetInterface(ptrVoE);

webrtc::VoENetwork* ptrVoENetwork = NULL;

ptrVoENetwork = webrtc::VoENetwork::GetInterface(ptrVoE);

webrtc::VoEFile* ptrVoEFile = NULL;

ptrVoEFile = webrtc::VoEFile::GetInterface(ptrVoE);

webrtc::VoEHardware* ptrVoEHardware = NULL;

ptrVoEHardware = webrtc::VoEHardware::GetInterface(ptrVoE);

然后可以选择设置tracefile的路径，这里我们还会对麦克风以及回放的声音做一个录制，所以也一并指明路径。

//Set Trace File and Record File

const std::string trace_filename = "webrtc_trace.txt";

VoiceEngine::SetTraceFilter(kTraceAll);

error = VoiceEngine::SetTraceFile(trace_filename.c_str());

if (error != )

{

    printf("ERROR in VoiceEngine::SetTraceFile\n");

    return error;

}

error = VoiceEngine::SetTraceCallback(NULL);

if (error != )

{

    printf("ERROR in VoiceEngine::SetTraceCallback\n");

    return error;

}

const std::string play_filename = "recorded_playout.wav";

const std::string mic_filename = "recorded_mic.wav";

接下来是初始化，获取VoiceEngine的版本号

//Init

error = ptrVoEBase->Init();

if (error != )

{

    printf("ERROR in VoEBase::Init\n");

    return error;

}

error = ptrVoEBase->RegisterVoiceEngineObserver(my_observer);

if (error != )

{

    printf("ERROR in VoEBase:;RegisterVoiceEngineObserver\n");

    return error;

}

printf("Version\n");

char tmp[];

error = ptrVoEBase->GetVersion(tmp);

if (error != )

{

    printf("ERROR in VoEBase::GetVersion\n");

    return error;

}

printf("%s\n", tmp);

这里同时还注册了一个VoiceEngineObserver对象，可以根据相应的error code输出信息，比如当检测到键盘敲击的噪音时会给出提示。这个类的定义如下：

class MyObserver : public VoiceEngineObserver

{

public:

    virtual void CallbackOnError(int channel, int err_code);

};

void MyObserver::CallbackOnError(int channel, int err_code)

{

    // Add printf for other error codes here

    if (err_code == VE_TYPING_NOISE_WARNING)

    {

        printf("  TYPING NOISE DETECTED \n");

    }

    else if (err_code == VE_TYPING_NOISE_OFF_WARNING)

    {

        printf("  TYPING NOISE OFF DETECTED \n");

    }

    else if (err_code == VE_RECEIVE_PACKET_TIMEOUT)

    {

        printf("  RECEIVE PACKET TIMEOUT \n");

    }

    else if (err_code == VE_PACKET_RECEIPT_RESTARTED)

    {

        printf("  PACKET RECEIPT RESTARTED \n");

    }

    else if (err_code == VE_RUNTIME_PLAY_WARNING)

    {

        printf("  RUNTIME PLAY WARNING \n");

    }

    else if (err_code == VE_RUNTIME_REC_WARNING)

    {

        printf("  RUNTIME RECORD WARNING \n");

    }

    else if (err_code == VE_SATURATION_WARNING)

    {

        printf("  SATURATION WARNING \n");

    }

    else if (err_code == VE_RUNTIME_PLAY_ERROR)

    {

        printf("  RUNTIME PLAY ERROR \n");

    }

    else if (err_code == VE_RUNTIME_REC_ERROR)

    {

        printf("  RUNTIME RECORD ERROR \n");

    }

    else if (err_code == VE_REC_DEVICE_REMOVED)

    {

        printf("  RECORD DEVICE REMOVED \n");

    }

}

以上完成了前期准备的工作，下面首先开始对网络的设置。如果是在本机上进行测试的话，ip地址直接写127.0.0.1即可，同时要注意的是，remote port和local port要一致。

//Network Settings

int audiochannel;

audiochannel = ptrVoEBase->CreateChannel();

if (audiochannel < )

{

    printf("ERROR in VoEBase::CreateChannel\n");

    return audiochannel;

}

VoiceChannelTransport* voice_channel_transport = new VoiceChannelTransport(ptrVoENetwork, audiochannel);

char ip[] = "127.0.0.1";

int rPort = ;//remote port

int lPort = ;//local port

error = voice_channel_transport->SetSendDestination(ip, rPort);

if (error != )

{

    printf("ERROR in set send ip and port\n");

    return error;

}

error = voice_channel_transport->SetLocalReceiver(lPort);

if (error != )

{

    printf("ERROR in set receiver and port\n");

    return error;

}

上面出现的VoiceChannelTransport类的定义如下

// Helper class for VoiceEngine tests.

class VoiceChannelTransport : public webrtc::test::UdpTransportData

{

public:

    VoiceChannelTransport(VoENetwork* voe_network, int channel);

    virtual ~VoiceChannelTransport();

    // Start implementation of UdpTransportData.

    void IncomingRTPPacket(const int8_t* incoming_rtp_packet,

        const size_t packet_length,

        const char* /*from_ip*/,

        const uint16_t /*from_port*/) override;

    void IncomingRTCPPacket(const int8_t* incoming_rtcp_packet,

        const size_t packet_length,

        const char* /*from_ip*/,

        const uint16_t /*from_port*/) override;

    // End implementation of UdpTransportData.

    // Specifies the ports to receive RTP packets on.

    int SetLocalReceiver(uint16_t rtp_port);

    // Specifies the destination port and IP address for a specified channel.

    int SetSendDestination(const char* ip_address, uint16_t rtp_port);

private:

    int channel_;

    VoENetwork* voe_network_;

    webrtc::test::UdpTransport* socket_transport_;

};

VoiceChannelTransport::VoiceChannelTransport(VoENetwork* voe_network,

    int channel)

    : channel_(channel),

    voe_network_(voe_network)

{

    uint8_t socket_threads = ;

    socket_transport_ = webrtc::test::UdpTransport::Create(channel, socket_threads);

    int registered = voe_network_->RegisterExternalTransport(channel, *socket_transport_);

#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)

    if (registered != )

        return;

#else

    assert(registered == );

#endif

}

VoiceChannelTransport::~VoiceChannelTransport()

{

    voe_network_->DeRegisterExternalTransport(channel_);

    webrtc::test::UdpTransport::Destroy(socket_transport_);

}

void VoiceChannelTransport::IncomingRTPPacket(

    const int8_t* incoming_rtp_packet,

    const size_t packet_length,

    const char* /*from_ip*/,

    const uint16_t /*from_port*/)

{

    voe_network_->ReceivedRTPPacket(channel_, incoming_rtp_packet, packet_length, PacketTime());

}

void VoiceChannelTransport::IncomingRTCPPacket(

    const int8_t* incoming_rtcp_packet,

    const size_t packet_length,

    const char* /*from_ip*/,

    const uint16_t /*from_port*/)

{

    voe_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet, packet_length);

}

int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port)

{

    static const int kNumReceiveSocketBuffers = ;

    int return_value = socket_transport_->InitializeReceiveSockets(this, rtp_port);

    if (return_value == )

    {

        return socket_transport_->StartReceiving(kNumReceiveSocketBuffers);

    }

    return return_value;

}

int VoiceChannelTransport::SetSendDestination(const char* ip_address, uint16_t rtp_port)

{

    return socket_transport_->InitializeSendSockets(ip_address, rtp_port);

}

完成了网络的设置后，进行编解码器的设置。这里简单的由用户选择使用哪一个编码器，当然还可以进一步对编码器的参数进行设置

//Setup Codecs

CodecInst codec_params;

CodecInst cinst;

for (int i = ; i < ptrVoECodec->NumOfCodecs(); ++i)

{

    int error = ptrVoECodec->GetCodec(i, codec_params);

    if (error != )

    {

        printf("ERROR in VoECodec::GetCodec\n");

        return error;

    }

    printf("%2d. %3d  %s/%d/%d \n", i, codec_params.pltype, codec_params.plname, codec_params.plfreq, codec_params.channels);

}

printf("Select send codec: ");

int codec_selection;

scanf("%i", &codec_selection);

ptrVoECodec->GetCodec(codec_selection, cinst);

error = ptrVoECodec->SetSendCodec(audiochannel, cinst);

if (error != )

{

    printf("ERROR in VoECodec::SetSendCodec\n");

    return error;

}

接下来进行录制设备和播放设备的设置

//Setup Devices

int rd(-), pd(-);

error = ptrVoEHardware->GetNumOfRecordingDevices(rd);

if (error != )

{

    printf("ERROR in VoEHardware::GetNumOfRecordingDevices\n");

    return error;

}

error = ptrVoEHardware->GetNumOfPlayoutDevices(pd);

if (error != )

{

    printf("ERROR in VoEHardware::GetNumOfPlayoutDevices\n");

    return error;

}

char dn[] = {  };

char guid[] = {  };

printf("\nPlayout devices (%d): \n", pd);

for (int j = ; j < pd; ++j)

{

    error = ptrVoEHardware->GetPlayoutDeviceName(j, dn, guid);

    if (error != )

    {

        printf("ERROR in VoEHardware::GetPlayoutDeviceName\n");

        return error;

    }

    printf("  %d: %s \n", j, dn);

}

printf("Recording devices (%d): \n", rd);

for (int j = ; j < rd; ++j)

{

    error = ptrVoEHardware->GetRecordingDeviceName(j, dn, guid);

    if (error != )

    {

        printf("ERROR in VoEHardware::GetRecordingDeviceName\n");

        return error;

    }

    printf("  %d: %s \n", j, dn);

}

printf("Select playout device: ");

scanf("%d", &pd);

error = ptrVoEHardware->SetPlayoutDevice(pd);

if (error != )

{

    printf("ERROR in VoEHardware::SetPlayoutDevice\n");

    return error;

}

printf("Select recording device: ");

scanf("%d", &rd);

getchar();

error = ptrVoEHardware->SetRecordingDevice(rd);

if (error != )

{

    printf("ERROR in VoEHardware::SetRecordingDevice\n");

    return error;

}

然后对音频预处理功能进行设置，这里作为示例，把各种预处理功能都enable了

//Audio Processing

error = ptrVoECodec->SetVADStatus(, );//FIX:why not use audio channel

if (error != )

{

    printf("ERROR in VoECodec::SetVADStatus\n");

    return error;

}

error = ptrVoEAp->SetAgcStatus();

if (error != )

{

    printf("ERROR in VoEAudioProcess::SetAgcStatus\n");

    return error;

}

error = ptrVoEAp->SetEcStatus();

if (error != )

{

    printf("ERROR in VoEAudioProcess::SetEcStatus\n");

    return error;

}

error = ptrVoEAp->SetNsStatus();

if (error != )

{

    printf("ERROR in VoEAudioProcess::SetNsStatus\n");

    return error;

}

error = ptrVoEAp->SetRxAgcStatus(audiochannel, );

if (error != )

{

    printf("ERROR in VoEAudioProcess::SetRxAgcStatus\n");

    return error;

}

error = ptrVoEAp->SetRxNsStatus(audiochannel, );

if (error != )

{

    printf("ERROR in VoEAudioProcess::SetRxNsStatus\n");

    return error;

}

至此，就可以开始发送、接收、录制了

//Start Receive

error = ptrVoEBase->StartReceive(audiochannel);

if (error != )

{

    printf("ERROR in VoEBase::StartReceive\n");

    return error;

}

//Start Playout

error = ptrVoEBase->StartPlayout(audiochannel);

if (error != )

{

    printf("ERROR in VoEBase::StartPlayout\n");

    return error;

}

//Start Send

error = ptrVoEBase->StartSend(audiochannel);

if (error != )

{

    printf("ERROR in VoEBase::StartSend\n");

    return error;

}

//Start Record

error = ptrVoEFile->StartRecordingMicrophone(mic_filename.c_str());

if (error != )

{

    printf("ERROR in VoEFile::StartRecordingMicrophone\n");

    return error;

}

error = ptrVoEFile->StartRecordingPlayout(audiochannel, play_filename.c_str());

if (error != )

{

    printf("ERROR in VoEFile::StartRecordingPlayout\n");

    return error;

}

在通话结束之后，还需要进行相应的stop\release

//Stop Record

error = ptrVoEFile->StopRecordingMicrophone();

if (error != )

{

    printf("ERROR in VoEFile::StopRecordingMicrophone\n");

    return error;

}

error = ptrVoEFile->StopRecordingPlayout(audiochannel);

if (error != )

{

    printf("ERROR in VoEFile::StopRecordingPlayout\n");

    return error;

}

//Stop Receive

error = ptrVoEBase->StopReceive(audiochannel);

if (error != )

{

    printf("ERROR in VoEBase::StopReceive\n");

    return error;

}

//Stop Send

error = ptrVoEBase->StopSend(audiochannel);

if (error != )

{

    printf("ERROR in VoEBase::StopSend\n");

    return error;

}

//Stop Playout

error = ptrVoEBase->StopPlayout(audiochannel);

if (error != )

{

    printf("ERROR in VoEBase::StopPlayout\n");

    return error;

}

//Delete Channel

error = ptrVoEBase->DeleteChannel(audiochannel);

if (error != )

{

    printf("ERROR in VoEBase::DeleteChannel\n");

    return error;

}

delete voice_channel_transport;

ptrVoEBase->DeRegisterVoiceEngineObserver();

error = ptrVoEBase->Terminate();

if (error != )

{

    printf("ERROR in VoEBase::Terminate\n");

    return error;

}

int remainingInterfaces = ;

remainingInterfaces += ptrVoEBase->Release();

remainingInterfaces = ptrVoECodec->Release();

remainingInterfaces += ptrVoEVolume->Release();

remainingInterfaces += ptrVoEFile->Release();

remainingInterfaces += ptrVoEAp->Release();

remainingInterfaces += ptrVoEHardware->Release();

remainingInterfaces += ptrVoENetwork->Release();

/*if (remainingInterfaces > 0)

{

printf("ERROR: Could not release all interfaces\n");

return -1;

}*/

bool deleted = webrtc::VoiceEngine::Delete(ptrVoE);

if (deleted == false)

{

    printf("ERROR in VoiceEngine::Delete\n");

    return -;

}

需要注意的是，这里remainingInterfaces最后不会为0，因为我们没有用到VoiceEngine的全部sub-apis。

至此，就实现了一个音频通话的功能。本项目源代码下载地址。github地址。

原文转自 http://blog.csdn.net/nonmarking/article/details/50577860

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