如何在Raspberry Pi上用C语言将接收到的UDP音频数据正确写入ALSA

我有2个Raspberry Pis,其中1个将UDP帧的音频数据传输到另一个Raspberry Pi.收到的UDP数据包各为160字节.传输的Raspberry Pi正在发送8KHz 8位单声道样本.接收Raspberry Pi使用带有QUDPSocket的Qt 5.4.0并尝试使用ALSA播放接收的数据.代码如下.每当字节到达接收Raspberry Pi时触发“readyRead”信号,缓冲区就会写入ALSA.我在接收Pi上的耳机插孔中发出了非常低声和毛躁的声音 – 但它是可识别的.所以它工作但听起来很糟糕.

>对于ALSA,我的配置有什么明显的错误吗？
>我应该如何使用snd_pcm_writei将收到的UDP数据包写入ALSA？

谢谢你的任何建议.

UdpReceiver::UdpReceiver(QObject *parent) : QObject(parent)
{

    // Debug
    qDebug() << "Setting up a UDP Socket...";

    // Create a socket
    m_Socket = new QUdpSocket(this);

    // Bind to the 2616 port
    bool didBind = m_Socket->bind(QHostAddress::Any,0x2616);
    if ( !didBind ) {
        qDebug() << "Error - could not bind to UDP Port!";
    }
    else {
        qDebug() << "Success binding to port 0x2616!";
    }

    // Get notified that data is incoming to the socket
    connect(m_Socket,SIGNAL(readyRead()),this,SLOT(readyRead()));

    // Init to Zero
    m_NumberUDPPacketsReceived = 0;

}

void UdpReceiver::readyRead() {

    // When data comes in
    QByteArray buffer;
    buffer.resize(m_Socket->pendingDatagramSize());

    QHostAddress sender;
    quint16 senderPort;

    // Cap buffer size
    int lenToRead = buffer.size();
    if ( buffer.size() > NOMINAL_AUDIO_BUFFER_SIZE ) {
        lenToRead = NOMINAL_AUDIO_BUFFER_SIZE;
    }

    // Read the data from the UDP Port
    m_Socket->readDatagram(buffer.data(),lenToRead,&sender,&senderPort);

    // Kick off audio playback
    if ( m_NumberUDPPacketsReceived == 0 ) {

        qDebug() << "Received Data - Setting up ALSA Now....";

        // Error handling
        int err;

        // Device to Write to
        char *snd_device_out  = "hw:0,0";

        if ((err = snd_pcm_open (&playback_handle,snd_device_out,SND_PCM_STREAM_PLAYBACK,0)) < 0) {
            fprintf (stderr,"cannot open audio device %s (%s)\n",snd_strerror (err));
            exit (1);
        }

        if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
            fprintf (stderr,"cannot allocate hardware parameter structure (%s)\n",snd_strerror (err));
            exit (1);
        }

        if ((err = snd_pcm_hw_params_any (playback_handle,hw_params)) < 0) {
            fprintf (stderr,"cannot initialize hardware parameter structure (%s)\n",snd_strerror (err));
            exit (1);
        }

        if ((err = snd_pcm_hw_params_set_access (playback_handle,hw_params,SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
            fprintf (stderr,"cannot set access type (%s)\n",snd_strerror (err));
            exit (1);
        }

        if ((err = snd_pcm_hw_params_set_format (playback_handle,SND_PCM_FORMAT_U8)) < 0) { // Unsigned 8 bit
            fprintf (stderr,"cannot set sample format (%s)\n",snd_strerror (err));
            exit (1);

        }

        uint sample_rate = 8000;
        if ((err = snd_pcm_hw_params_set_rate (playback_handle,sample_rate,0)) < 0) { // 8 KHz
            fprintf (stderr,"cannot set sample rate (%s)\n",snd_strerror (err));
            exit (1);
        }

        if ((err = snd_pcm_hw_params_set_channels (playback_handle,1)) < 0) { // 1 Channel Mono
            fprintf (stderr,"cannot set channel count (%s)\n",snd_strerror (err));
            exit (1);
        }

        if ((err = snd_pcm_hw_params (playback_handle,"cannot set parameters (%s)\n",snd_strerror (err));
            exit (1);
        }

        snd_pcm_hw_params_free (hw_params);

        // Flush handle prepare for playback
        snd_pcm_drop(playback_handle);

        if ((err = snd_pcm_prepare (playback_handle)) < 0) {
            fprintf (stderr,"cannot prepare audio interface for use (%s)\n",snd_strerror (err));
            exit (1);
        }

        qDebug() << "Done Setting up ALSA....";

    }

    // Grab the buffer
    m_Buffer = buffer.data();

    // Write the data to the ALSA device
    int error;
    if ((error = snd_pcm_writei (playback_handle,m_Buffer,NOMINAL_AUDIO_BUFFER_SIZE)) != NOMINAL_AUDIO_BUFFER_SIZE) {
        fprintf (stderr,"write to audio interface failed (%s)\n",snd_strerror (error));
        exit (1);
    }

    // Count up
    m_NumberUDPPacketsReceived++;

}

解决方法

我不理解代码中的“封顶缓冲区大小”部分.如果传入的数据大于您的任意NOMINAL_AUDIO_BUFFER_SIZE,那么您将切断传入的数据.你试过删除那段代码吗？