配置ALSA接收和播放Raspberry Pi上的原始PCM模拟输出

Question

我已经使用PulseAudio和Qt5.4的QAudioOutput在Raspberry Pi上的3.5mm耳机插孔模拟输出上成功播放了音频。音频通过8位采样率的8KHz的XBee链路从远端麦克风成功传输。

PulseAudio有一个巨大的延迟，所以我决定链接到libasound（ALSA）并直接播放音频。我的代码在下面，并成功打开并播放声音，但几乎无法辨认，有很多crack啪声和吱吱声。如果我对着远端麦克风说话，我很快就会听到从Pi发出的高音划伤和吱吱声（但它不是很好的音频）。我想我有我的参数搞砸了。

1.）数据以BigEndian传输 - QAudioOutput允许您通知它样本是BigEndian。但这些是U8样本，我需要担心排序吗？ 2.）你能看到下面我的配置有什么问题吗？ 3.）如何找出Pi上输出的ALSA的片段大小？ 4.）有人可以解释我应该如何将我的缓冲区写入音频设备？

谢谢！

这里是我的代码：

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_device_out, 
        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, hw_params, 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_near (playback_handle, hw_params, &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, hw_params, 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, hw_params)) < 0) { 
      fprintf (stderr, "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; 
    for (int i = 0; i < 10; ++i) { 
     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++; 

} 

Answer 1

1. 号
2. snd_pcm_hw_params_set_rate_near()将改变率最接近的支持率。你可能不希望这样。

3. ALSA没有片段大小。

   它具有缓冲区和周期大小;您需要根据您的时间要求进行设置（请参阅ALSA: Relation between period size of speaker and mic）。

4. 你不能简单地输出收到的样本;您必须将它们重新采样到播放设备的速度（即使它使用相同的标称采样率，也不完全相同）。