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XBee链路以8KHz成功从远程麦克风传输，带有8位采样。

PulseAudio有一个巨大的延迟，所以我决定链接libasound（ALSA）并直接播放音频。我的代码在下方并成功打开并播放声音，但它几乎无法识别，有很多噼啪声和吱吱声。如果我对着遥控话筒说话，我很快就会听到Pi从耳机中听到抬高的划痕和吱吱声（但这不是很好的音频）。我想我的参数搞砸了。

1。）数据在BigEndian中传输 - QAudioOutput允许您通知它样本是BigEndian。但这些是U8样本，所以我需要担心字节序吗？ 2.）你能看到我的配置有什么问题吗？ 3.）如何为Pi上的输出计算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吗？

谢谢！

这是我的代码：

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

没有
snd_pcm_hw_params_set_rate_near()会将费率更改为最接近的支持费率。你可能不想要那个。
ALSA没有片段大小。

它有缓冲区和周期大小;您需要根据时间要求进行设置（参见ALSA: Relation between period size of speaker and mic）。
您不能简单地输出收到的样本;你必须将它们重新取样到播放设备的速度（即使它使用相同的标称采样率，它也不完全相同）。

ALSA在Raspberry Pi模拟输出上接收和播放原始PCM的配置

1 个答案: