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

How to Properly Write Received UDP Audio Data to ALSA with C++ on Raspberry Pi

本文关键字:数据 音频 UDP ALSA Raspberry Pi 上用 C++      更新时间:2023-10-16

我有两个树莓派,其中一个将音频数据的UDP帧传输到另一个树莓皮。接收到的UDP数据包每个为160字节。发送Raspberry Pi正在发送8KHz 8位Mono样本。接收Raspberry Pi使用Qt 5.4.0和QUDPSocket,并尝试用ALSA播放接收到的数据。代码如下。当字节到达接收Raspberry Pi时,每次触发"readyRead"信号时,缓冲区都会写入ALSA。接收Pi上的耳机插孔发出的声音非常清脆,但很容易辨认。所以它是有效的,但听起来很可怕。

  1. 我下面的ALSA配置有什么明显的错误吗
  2. 我应该如何使用snd_pcm_writei将接收到的UDP数据包写入ALSA

谢谢你的建议。

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_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 (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;
    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++;
}

我不理解代码中的"限制缓冲区大小"部分。如果传入数据大于您任意的NOMINAL_AUDIO_BUFFER_SIZE,则您正在截断该传入数据。你试过删除那部分代码吗?