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

I have 2 Raspberry Pis and 1 of them transmits UDP frames of Audio data to the other Raspberry Pi. The UDP Packets received are 160 Bytes each. The transmitting Raspberry Pi is sending 8KHz 8-bit Mono samples. The receiving Raspberry Pi uses Qt 5.4.0 with QUDPSocket and tries to play the received data with ALSA. The code is below. Each time the "readyRead " signal is fired when bytes arrive on the receiving Raspberry Pi, the Buffer is written to ALSA. I have very Choppy and Glitchy sound coming out of the headphone Jack on the Receiving Pi - but it is recognizable. So it is working but sounds Terrible.
Is there anything glaringly wrong with my configuration below for ALSA?
How should I approach writing the received UDP packets to ALSA with snd_pcm_writei?
Thank you for any suggestions.
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++;}
Can you play sounds better when using another program? The jack sound stuff (but not HDMI) of some(?) models just is terrible itself, independent from the software. –  deviantfan Feb 8 at 23:50 
 
Thanks for the input. But if I use ALSA utilities' "aplay" program then the Audio output sounds great. I'm pretty sure this is a configuration problem. –  PhilBot Feb 9 at 0:29 
 
You still are not setting buffer/period sizes. –  CL. Feb 9 at 8:17
 
You should process the incoming datagrams with a while inside the SLOT, as done for instance here to ensure that all the available data is processed with the current signal. –  BaCaRoZzo Feb 9 at 10:38 
1 
Set up your audio buffer to gather 1 full second and then play in chunks of 1 second buffers. See if quality gets better (it should). Change the design to have a listening thread that reads data and fills a buffer while another thread plays it with a delay of, let's say, 0.1 seconds by continuously feeding data. You have 8000 samples per second. 5 UDP messages means 800 samples, which is 0.1 seconds. So you can start with this. By feeding 160 samples in a run you play 0.02 seconds at a time. Then you stop playing, and on the same thread you read another UDP message. –  Bogdan V. Feb 18 at 9:48