Configuration for ALSA Receiving and Playing Raw PCM on Raspberry Pi Analogue Output

Question

I have successfully played audio on the 3.5mm headphone jack analogue output on the Raspberry Pi using PulseAudio and Qt5.4's QAudioOutput. The audio is successfully being transported from a remote microphone over a XBee link at 8KHz with 8-bit samples.

There was a huge latency with PulseAudio so I decided to link against libasound ( ALSA ) and play the audio directly. My code is below and successfully opens and plays sound but it is almost unrecognizable, there is lots of crackling and squeaking. If I speak into the remote microphone, I very quickly hear elevated scratching and squeaking in the headphones from the Pi ( but it is not good audio ) . I think I have my parameters messed up.

1.) The data is transmitted in BigEndian - QAudioOutput allows you to inform it that the samples are BigEndian. But these are U8 samples so do I need to worry about endianness? 2.) Can you see anything wrong with my configuration below? 3.) How do I figure out Fragment Size for ALSA for the output on the Pi? 4.) Can someone explain how I should write my buffer to the audio device?

Thanks!

Here is my code:

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++;

}

CL. CL. · Accepted Answer · 2015-02-07T22:32:02

No.
snd_pcm_hw_params_set_rate_near() will change the rate to the nearest supported rate. You probably don't want that.
ALSA has no fragment size.

It has buffer and period sizes; you need to set them according to your timing requirements (see ALSA: Relation between period size of speaker and mic).
You cannot simply output the received samples; you must resample them to the speed of the playback device (which is not exactly the same even if it uses the same nominal sample rate).

Configuration for ALSA Receiving and Playing Raw PCM on Raspberry Pi Analogue Output

1 Answers