void
audio_init()
{
  unsigned int buffer_time = 50000;
  const char* device;
  int err;

  if(audio_initialised)
    return;

  audio_initialised = 1;

  device = getenv("ALSA_DEVICE");

  if(!device)
    device = "default";

  if(0 > (err = snd_pcm_open(&playback_handle, device,
                             SND_PCM_STREAM_PLAYBACK, 0/*SND_PCM_NONBLOCK*/)))
    errx(EXIT_FAILURE, "Audio: Cannot open device %s: %s", device, snd_strerror(err));

  if(0 > (err = snd_pcm_sw_params_malloc(&sw_params)))
    errx(EXIT_FAILURE, "Audio: Could not allocate software parameter structure: %s",
         snd_strerror(err));

  if(0 > (err = snd_pcm_hw_params_malloc(&hw_params)))
    errx(EXIT_FAILURE, "Audio: Could not allocate hardware parameter structure: %s",
         snd_strerror(err));

  if(0 > (err = snd_pcm_hw_params_any(playback_handle, hw_params)))
    errx(EXIT_FAILURE, "Audio: Could not initializa hardware parameters: %s",
         snd_strerror(err));

  if(0 > (err = snd_pcm_hw_params_set_access(playback_handle, hw_params,
                                             SND_PCM_ACCESS_RW_INTERLEAVED)))
    errx(EXIT_FAILURE, "Audio: Could not set access type: %s", snd_strerror(err));

  if(0 > (err = snd_pcm_hw_params_set_format(playback_handle, hw_params,
                                             SND_PCM_FORMAT_S16)))
    errx(EXIT_FAILURE, "Audio: Could not set sample format to signed 16 bit "
         "native endian: %s", snd_strerror(err));

  if(0 > (err = snd_pcm_hw_params_set_rate_near(playback_handle, hw_params,
                                                &rate, 0)))
    errx(EXIT_FAILURE, "Audio: Could not set sample rate %uHz: %s", rate,
         snd_strerror(err));

  if(0 > (err = snd_pcm_hw_params_set_channels(playback_handle, hw_params, 2)))
    errx(EXIT_FAILURE, "Audio: Could not set channel count to %u: %s",
         2, snd_strerror(err));

  snd_pcm_hw_params_set_buffer_time_near(playback_handle, hw_params, &buffer_time, 0);

  if(0 > (err = snd_pcm_hw_params(playback_handle, hw_params)))
    errx(EXIT_FAILURE, "Audio: Could not set hardware parameters: %s", snd_strerror(err));

  fprintf(stderr, "Buffer time is %.3f seconds\n", buffer_time / 1.0e6);

  if(0 > (err = snd_pcm_sw_params_current(playback_handle, sw_params)))
    errx(EXIT_FAILURE, "Audio: Could not initialise software parameters: %s",
         snd_strerror(err));

  snd_pcm_sw_params_set_start_threshold(playback_handle, sw_params, 0);
  snd_pcm_sw_params_set_avail_min(playback_handle, sw_params, 1024);

  snd_pcm_uframes_t min;
  snd_pcm_sw_params_get_avail_min(sw_params, &min);
  fprintf(stderr, "Minimum %u\n", (unsigned) min);

  if(0 > (err = snd_pcm_sw_params(playback_handle, sw_params)))
    errx(EXIT_FAILURE, "Audio: Could not set software parameters: %s",
        snd_strerror(err));

  buffer_size = snd_pcm_avail_update(playback_handle);
}

int initAlsa(char **argv,int optind)
{
    snd_pcm_hw_params_t *hw_params;
    int err,n;
    unsigned int Fs;

    if ((err = snd_pcm_open(&capture_handle, argv[optind],
                            SND_PCM_STREAM_CAPTURE, 0)) < 0) {
        fprintf(stderr, "Alsa cannot open audio device %s (%s)\n",argv[optind], snd_strerror(err));
        return 1;
    }

    if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
        fprintf(stderr,
                "Alsa cannot allocate hardware parameter structure (%s)\n",snd_strerror(err));
        return 1;
    }

    if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) {
        fprintf(stderr,
                "Alsa cannot initialize hardware parameter structure (%s)\n",snd_strerror(err));
        return 1;
    }

    if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params,SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
        fprintf(stderr, "Alsa cannot set access type (%s)\n",snd_strerror(err));
        return 1;
    }

    if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params,SND_PCM_FORMAT_S16)) < 0) {
        fprintf(stderr, "Alsa cannot set sample format (%s)\n",snd_strerror(err));
        return 1;
    }

    snd_pcm_hw_params_set_rate_resample(capture_handle, hw_params,0);
    Fs=19200;
    n=1;
    if ((err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &Fs,&n)) < 0) {
        fprintf(stderr, "Alsa cannot set sample rate (%s)\n",snd_strerror(err));
        return 1;
    }
    fprintf(stderr, "Alsa sample rate %d\n",Fs);

    if(snd_pcm_hw_params_get_channels (hw_params, &nbch)!=0) {
        fprintf(stderr, "Alsa cannot get number of channels\n");
        return 1;
    }
    if(nbch>4) {
        fprintf(stderr, "Alsa too much channels\n");
        return 1;

    }
    if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) {
        fprintf(stderr, "Alsa cannot set parameters (%s)\n",snd_strerror(err));
        return 1;
    }
    snd_pcm_hw_params_free(hw_params);

    if ((err = snd_pcm_prepare(capture_handle)) < 0) {
        fprintf(stderr,
                "Alsa cannot prepare audio interface for use (%s)\n",snd_strerror(err));
        return 1;
    }

    for(n=0; n<nbch; n++) {
        channel[n].chn=n;
        channel[n].Infs=Fs;
        channel[n].InBuff=malloc(MAXNBFRAMES*sizeof(sample_t));
    }
    for(; n<MAXNBCHANNELS; n++) channel[n].Infs=0;

    return (0);
}

bool OutputALSA::initialize(quint32 freq, ChannelMap map, Qmmp::AudioFormat format)
{
    m_inited = false;

    if (pcm_handle)
        return false;

    if (snd_pcm_open(&pcm_handle, pcm_name, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK) < 0)
    {
        qWarning ("OutputALSA: Error opening PCM device %s", pcm_name);
        return false;
    }

    // we need to configure

    uint rate = freq; /* Sample rate */
    uint exact_rate = freq;   /* Sample rate returned by */

    /* load settings from config */
    QSettings settings(Qmmp::configFile(), QSettings::IniFormat);
    settings.beginGroup("ALSA");
    uint buffer_time = settings.value("buffer_time",500).toUInt()*1000;
    uint period_time = settings.value("period_time",100).toUInt()*1000;
    bool use_pause =  settings.value("use_snd_pcm_pause", false).toBool();
    settings.endGroup();

    snd_pcm_hw_params_t *hwparams = 0;
    snd_pcm_sw_params_t *swparams = 0;
    int err; //alsa error code

    //hw params
    snd_pcm_hw_params_alloca(&hwparams);
    if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
    {
        qWarning("OutputALSA: Can not read configuration for PCM device: %s", snd_strerror(err));
        return false;
    }
    if (m_use_mmap)
    {
        if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_MMAP_INTERLEAVED)) < 0)
        {
            qWarning("OutputALSA: Error setting mmap access: %s", snd_strerror(err));
            m_use_mmap = false;
        }
    }
    if (!m_use_mmap)
    {
        if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
        {
            qWarning("OutputALSA: Error setting access: %s", snd_strerror(err));
            return false;
        }
    }
    snd_pcm_format_t alsa_format = SND_PCM_FORMAT_UNKNOWN;
    switch (format)
    {
    case Qmmp::PCM_S8:
        alsa_format = SND_PCM_FORMAT_S8;
        break;
    case Qmmp::PCM_S16LE:
        alsa_format = SND_PCM_FORMAT_S16_LE;
        break;
    case Qmmp::PCM_S24LE:
        alsa_format = SND_PCM_FORMAT_S24_LE;
        break;
    case Qmmp::PCM_S32LE:
        alsa_format = SND_PCM_FORMAT_S32_LE;
        break;
    default:
        qWarning("OutputALSA: unsupported format detected");
        return false;
    }
    if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, alsa_format)) < 0)
    {
        qDebug("OutputALSA: Error setting format: %s", snd_strerror(err));
        return false;
    }
    exact_rate = rate;

    if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0)) < 0)
    {
        qWarning("OutputALSA: Error setting rate: %s", snd_strerror(err));
        return false;
    }
    if (rate != exact_rate)
    {
        qWarning("OutputALSA: The rate %d Hz is not supported by your hardware.\n==> Using %d Hz instead.", rate, exact_rate);
        rate = exact_rate;
    }
    uint c = map.count();
    if ((err = snd_pcm_hw_params_set_channels_near(pcm_handle, hwparams, &c)) < 0)
    {
        qWarning("OutputALSA: Error setting channels: %s", snd_strerror(err));
        return false;
    }
    if (c != (uint)map.count())
    {
        qWarning("OutputALSA: The channel number %d is not supported by your hardware", map.count());
        qWarning("==> Using %d instead.", c);
    }
//.........这里部分代码省略.........

int main(int argc, char *argv[]) {
    const char *dev;
    int r, cap, count = 0;
    snd_pcm_hw_params_t *hwparams;
    snd_pcm_sw_params_t *swparams;
    snd_pcm_status_t *status;
    snd_pcm_t *pcm;
    unsigned rate = SAMPLE_RATE;
    unsigned periods = 2;
    snd_pcm_uframes_t boundary, buffer_size = SAMPLE_RATE/10; /* 100s */
    int dir = 1;
    int fillrate;
    struct timespec start, last_timestamp = { 0, 0 };
    uint64_t start_us, last_us = 0;
    snd_pcm_sframes_t last_avail = 0, last_delay = 0;
    struct pollfd *pollfds;
    int n_pollfd;
    int64_t sample_count = 0;
    uint16_t *samples;
    struct sched_param sp;

    r = -1;
#ifdef _POSIX_PRIORITY_SCHEDULING
    sp.sched_priority = 5;
    r = pthread_setschedparam(pthread_self(), SCHED_RR, &sp);
#endif
    if (r)
        printf("Could not get RT prio. :(\n");

    snd_pcm_hw_params_alloca(&hwparams);
    snd_pcm_sw_params_alloca(&swparams);
    snd_pcm_status_alloca(&status);

    r = clock_gettime(CLOCK_MONOTONIC, &start);
    assert(r == 0);

    start_us = timespec_us(&start);

    dev = argc > 1 ? argv[1] : "front:0";
    cap = argc > 2 ? atoi(argv[2]) : 0;
    fillrate = argc > 3 ? atoi(argv[3]) : 1;

    samples = calloc(fillrate, CHANNELS*sizeof(uint16_t));
    assert(samples);

    if (cap == 0)
      r = snd_pcm_open(&pcm, dev, SND_PCM_STREAM_PLAYBACK, 0);
    else
      r = snd_pcm_open(&pcm, dev, SND_PCM_STREAM_CAPTURE, 0);
    assert(r == 0);

    r = snd_pcm_hw_params_any(pcm, hwparams);
    assert(r == 0);

    r = snd_pcm_hw_params_set_rate_resample(pcm, hwparams, 0);
    assert(r == 0);

    r = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
    assert(r == 0);

    r = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE);
    assert(r == 0);

    r = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, NULL);
    assert(r == 0);

    r = snd_pcm_hw_params_set_channels(pcm, hwparams, CHANNELS);
    assert(r == 0);

    r = snd_pcm_hw_params_set_periods_integer(pcm, hwparams);
    assert(r == 0);

    r = snd_pcm_hw_params_set_periods_near(pcm, hwparams, &periods, &dir);
    assert(r == 0);

    r = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, &buffer_size);
    assert(r == 0);

    r = snd_pcm_hw_params(pcm, hwparams);
    assert(r == 0);

    r = snd_pcm_hw_params_current(pcm, hwparams);
    assert(r == 0);

    r = snd_pcm_sw_params_current(pcm, swparams);
    assert(r == 0);

    if (cap == 0)
      r = snd_pcm_sw_params_set_avail_min(pcm, swparams, 1);
    else
      r = snd_pcm_sw_params_set_avail_min(pcm, swparams, 0);
    assert(r == 0);

    r = snd_pcm_sw_params_set_period_event(pcm, swparams, 0);
    assert(r == 0);

    r = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
    assert(r == 0);
    r = snd_pcm_sw_params_set_start_threshold(pcm, swparams, buffer_size - (buffer_size % fillrate));
    assert(r == 0);
//.........这里部分代码省略.........

static void
alsa_configure (struct sound_device *sd)
{
  int val, err, dir;
  unsigned uval;
  struct alsa_params *p = (struct alsa_params *) sd->data;
  snd_pcm_uframes_t buffer_size;

  eassert (p->handle != 0);

  err = snd_pcm_hw_params_malloc (&p->hwparams);
  if (err < 0)
    alsa_sound_perror ("Could not allocate hardware parameter structure", err);

  err = snd_pcm_sw_params_malloc (&p->swparams);
  if (err < 0)
    alsa_sound_perror ("Could not allocate software parameter structure", err);

  err = snd_pcm_hw_params_any (p->handle, p->hwparams);
  if (err < 0)
    alsa_sound_perror ("Could not initialize hardware parameter structure", err);

  err = snd_pcm_hw_params_set_access (p->handle, p->hwparams,
                                      SND_PCM_ACCESS_RW_INTERLEAVED);
  if (err < 0)
    alsa_sound_perror ("Could not set access type", err);

  val = sd->format;
  err = snd_pcm_hw_params_set_format (p->handle, p->hwparams, val);
  if (err < 0)
    alsa_sound_perror ("Could not set sound format", err);

  uval = sd->sample_rate;
  err = snd_pcm_hw_params_set_rate_near (p->handle, p->hwparams, &uval, 0);
  if (err < 0)
    alsa_sound_perror ("Could not set sample rate", err);

  val = sd->channels;
  err = snd_pcm_hw_params_set_channels (p->handle, p->hwparams, val);
  if (err < 0)
    alsa_sound_perror ("Could not set channel count", err);

  err = snd_pcm_hw_params (p->handle, p->hwparams);
  if (err < 0)
    alsa_sound_perror ("Could not set parameters", err);


  err = snd_pcm_hw_params_get_period_size (p->hwparams, &p->period_size, &dir);
  if (err < 0)
    alsa_sound_perror ("Unable to get period size for playback", err);

  err = snd_pcm_hw_params_get_buffer_size (p->hwparams, &buffer_size);
  if (err < 0)
    alsa_sound_perror ("Unable to get buffer size for playback", err);

  err = snd_pcm_sw_params_current (p->handle, p->swparams);
  if (err < 0)
    alsa_sound_perror ("Unable to determine current swparams for playback",
                       err);

  /* Start the transfer when the buffer is almost full */
  err = snd_pcm_sw_params_set_start_threshold (p->handle, p->swparams,
                                               (buffer_size / p->period_size)
                                               * p->period_size);
  if (err < 0)
    alsa_sound_perror ("Unable to set start threshold mode for playback", err);

  /* Allow the transfer when at least period_size samples can be processed */
  err = snd_pcm_sw_params_set_avail_min (p->handle, p->swparams, p->period_size);
  if (err < 0)
    alsa_sound_perror ("Unable to set avail min for playback", err);

  err = snd_pcm_sw_params (p->handle, p->swparams);
  if (err < 0)
    alsa_sound_perror ("Unable to set sw params for playback\n", err);

  snd_pcm_hw_params_free (p->hwparams);
  p->hwparams = NULL;
  snd_pcm_sw_params_free (p->swparams);
  p->swparams = NULL;

  err = snd_pcm_prepare (p->handle);
  if (err < 0)
    alsa_sound_perror ("Could not prepare audio interface for use", err);

  if (sd->volume > 0)
    {
      int chn;
      snd_mixer_t *handle;
      snd_mixer_elem_t *e;
      if (snd_mixer_open (&handle, 0) >= 0)
        {
	  char const *file = string_default (sd->file,
					     DEFAULT_ALSA_SOUND_DEVICE);
          if (snd_mixer_attach (handle, file) >= 0
              && snd_mixer_load (handle) >= 0
              && snd_mixer_selem_register (handle, NULL, NULL) >= 0)
            for (e = snd_mixer_first_elem (handle);
                 e;
                 e = snd_mixer_elem_next (e))
//.........这里部分代码省略.........

void *data_streaming(void *socket_desc)
{
  int rc;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val;
  int dir;
  snd_pcm_uframes_t frames;
  int i;

  /* Open PCM device for recording (capture). */
  rc = snd_pcm_open(&handle, "plughw:1,0",
                    SND_PCM_STREAM_CAPTURE, 0);
  if (rc < 0) {
    fprintf(stderr,
            "unable to open pcm device: %s\n",
            snd_strerror(rc));
    exit(1);
  }

  /* Allocate a hardware parameters object. */
  snd_pcm_hw_params_alloca(&params);

  /* Fill it in with default values. */
  snd_pcm_hw_params_any(handle, params);

  /* Set the desired hardware parameters. */

  /* Interleaved mode */
  snd_pcm_hw_params_set_access(handle, params,
                      SND_PCM_ACCESS_RW_INTERLEAVED);

  /* Signed 16-bit little-endian format */

  /* Signed 16-bit little-endian format */
  snd_pcm_hw_params_set_format(handle, params,
                              SND_PCM_FORMAT_U8);

  /* Two channels (stereo) */
  snd_pcm_hw_params_set_channels(handle, params, 1);

  /* 44100 bits/second sampling rate (CD quality) */
  val = 32768;
  snd_pcm_hw_params_set_rate_near(handle, params,
                                  &val, &dir);

  /* Set period size to 32 frames. */
  frames = SIZE;
  snd_pcm_hw_params_set_period_size_near(handle,
                              params, &frames, &dir);

  /* Write the parameters to the driver */
  rc = snd_pcm_hw_params(handle, params);
  if (rc < 0) {
    fprintf(stderr,
            "unable to set hw parameters: %s\n",
            snd_strerror(rc));
    exit(1);
  }

  /* Use a buffer large enough to hold one period */
  snd_pcm_hw_params_get_period_size(params,
                                      &frames, &dir);
 /* 2 bytes/sample, 2 channels */

  /* We want to loop for 5 seconds */
  snd_pcm_hw_params_get_period_time(params,
                                         &val, &dir);
  
  while (1) {
        pthread_mutex_lock(&mutex);
        rc = snd_pcm_readi(handle, buffer, frames);
        pthread_kill(tid, SIGUSR1);
        pthread_mutex_unlock(&mutex);

    if (rc == -EPIPE) {
      /* EPIPE means overrun */
      fprintf(stderr, "overrun occurred\n");
      snd_pcm_prepare(handle);
    } else if (rc < 0) {
      fprintf(stderr,"error from read: %s\n",snd_strerror(rc));
    } else if (rc != (int)frames) {
      fprintf(stderr, "short read, read %d frames\n", rc);
    }
   }

  snd_pcm_drain(handle);
  snd_pcm_close(handle);
  free(buffer);

    return 0;
}

static int
quh_alsa_config (st_quh_nfo_t *file)
{
  (void) file;
#if 0
  snd_pcm_hw_params_t *hw_params;
  snd_pcm_format_t format;
  int rate = 0;
  
  if (snd_pcm_hw_params_malloc (&hw_params) < 0)
    return -1;

  if (snd_pcm_hw_params_any (handle, hw_params) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  if (snd_pcm_hw_params_set_access (handle, hw_params,
                                    SND_PCM_ACCESS_RW_INTERLEAVED) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  switch (file->size)
    {
    case 1:
      format = SND_PCM_FORMAT_S8;
      break;
    case 2:
      format = SND_PCM_FORMAT_S16;
      break;
    case 3:
      format = SND_PCM_FORMAT_S24;
      break;
    default:
      format = SND_PCM_FORMAT_S16;
      break;
    }

  if (snd_pcm_hw_params_set_format (handle, hw_params, format) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  rate = file->rate;
  if (snd_pcm_hw_params_set_rate_near (handle, hw_params, rate, 0) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  if ((float) rate * 1.05 < file->rate || (float) rate * 0.95 > file->rate)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  if (snd_pcm_hw_params_set_channels (handle, hw_params, file->channels) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  if (snd_pcm_hw_params (handle, hw_params) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  snd_pcm_hw_params_free (hw_params);
#endif
  return 0;
}

alsa_dev_t* alsa_open(char *name)
{
    alsa_dev_t *alsa_dev = NULL;
    snd_pcm_hw_params_t *params = NULL;
    uint32_t val = 0;
    int32_t rc = 0;
    int32_t dir = 0;

    alsa_dev = malloc(sizeof(alsa_dev_t));
    if (!alsa_dev)
    {
        syslog(LOG_ERR, "Failed to create audio device: Out of memory");
        exit(EXIT_FAILURE);
    }

    /* Open PCM device for recording (capture). */
    rc = snd_pcm_open(&alsa_dev->snd_pcm, name, SND_PCM_STREAM_CAPTURE, 0);
    if (rc < 0)
    {
        syslog(LOG_ERR, "Cannot open pcm device: %s", snd_strerror(rc));
        exit(EXIT_FAILURE);
    }

    alsa_dev->name = name;

    /* Allocate a hardware parameters object. */
    snd_pcm_hw_params_alloca(&params);

    /* Fill it in with default values. */
    snd_pcm_hw_params_any(alsa_dev->snd_pcm, params);

    /* Set the desired hardware parameters. */

    /* Interleaved mode */
    snd_pcm_hw_params_set_access(alsa_dev->snd_pcm, params, 
                                 SND_PCM_ACCESS_RW_INTERLEAVED);

    /* Signed 16-bit little-endian format */
    snd_pcm_hw_params_set_format(alsa_dev->snd_pcm, params, 
                                 SND_PCM_FORMAT_S16_LE);

    /* Two channels (stereo) */
    snd_pcm_hw_params_set_channels(alsa_dev->snd_pcm, params, AUDIO_CHANNEL_NUM);

    /* 44100 bits/second sampling rate (CD quality) */
    val = AUDIO_SAMPLE_RATE;
    snd_pcm_hw_params_set_rate_near(alsa_dev->snd_pcm, params, &val, &dir);

    alsa_dev->frames = 1152;
    snd_pcm_hw_params_set_period_size_near(alsa_dev->snd_pcm, params, 
                                           &alsa_dev->frames, &dir);

    /* Write the parameters to the driver */
    rc = snd_pcm_hw_params(alsa_dev->snd_pcm, params);
    if (rc < 0)
    {
        syslog(LOG_ERR, "Cannot set hw parameters: %s", snd_strerror(rc));
        exit(EXIT_FAILURE);
    }

    /* Use a buffer large enough to hold one period */
    snd_pcm_hw_params_get_period_size(params, &alsa_dev->frames, &dir);

    /* We want to loop for 5 seconds */
    snd_pcm_hw_params_get_period_time(params, &val, &dir);

    snd_pcm_prepare(alsa_dev->snd_pcm);

    snd_pcm_start(alsa_dev->snd_pcm);

    return alsa_dev;
}

av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
                         unsigned int *sample_rate,
                         int channels, enum CodecID *codec_id)
{
    AlsaData *s = ctx->priv_data;
    const char *audio_device;
    int res, flags = 0;
    snd_pcm_format_t format;
    snd_pcm_t *h;
    snd_pcm_hw_params_t *hw_params;
    snd_pcm_uframes_t buffer_size, period_size;
    int64_t layout = ctx->streams[0]->codec->channel_layout;

    if (ctx->filename[0] == 0) audio_device = "default";
    else                       audio_device = ctx->filename;

    if (*codec_id == CODEC_ID_NONE)
        *codec_id = DEFAULT_CODEC_ID;
    format = codec_id_to_pcm_format(*codec_id);
    if (format == SND_PCM_FORMAT_UNKNOWN) {
        av_log(ctx, AV_LOG_ERROR, "sample format 0x%04x is not supported\n", *codec_id);
        return AVERROR(ENOSYS);
    }
    s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * channels;

    if (ctx->flags & AVFMT_FLAG_NONBLOCK) {
        flags = SND_PCM_NONBLOCK;
    }
    res = snd_pcm_open(&h, audio_device, mode, flags);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot open audio device %s (%s)\n",
               audio_device, snd_strerror(res));
        return AVERROR(EIO);
    }

    res = snd_pcm_hw_params_malloc(&hw_params);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot allocate hardware parameter structure (%s)\n",
               snd_strerror(res));
        goto fail1;
    }

    res = snd_pcm_hw_params_any(h, hw_params);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot initialize hardware parameter structure (%s)\n",
               snd_strerror(res));
        goto fail;
    }

    res = snd_pcm_hw_params_set_access(h, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set access type (%s)\n",
               snd_strerror(res));
        goto fail;
    }

    res = snd_pcm_hw_params_set_format(h, hw_params, format);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set sample format 0x%04x %d (%s)\n",
               *codec_id, format, snd_strerror(res));
        goto fail;
    }

    res = snd_pcm_hw_params_set_rate_near(h, hw_params, sample_rate, 0);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set sample rate (%s)\n",
               snd_strerror(res));
        goto fail;
    }

    res = snd_pcm_hw_params_set_channels(h, hw_params, channels);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set channel count to %d (%s)\n",
               channels, snd_strerror(res));
        goto fail;
    }

    snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
    buffer_size = FFMIN(buffer_size, ALSA_BUFFER_SIZE_MAX);
    /* TODO: maybe use ctx->max_picture_buffer somehow */
    res = snd_pcm_hw_params_set_buffer_size_near(h, hw_params, &buffer_size);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set ALSA buffer size (%s)\n",
               snd_strerror(res));
        goto fail;
    }

    snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, NULL);
    if (!period_size)
        period_size = buffer_size / 4;
    res = snd_pcm_hw_params_set_period_size_near(h, hw_params, &period_size, NULL);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set ALSA period size (%s)\n",
               snd_strerror(res));
        goto fail;
    }
    s->period_size = period_size;

    res = snd_pcm_hw_params(h, hw_params);
    if (res < 0) {
//.........这里部分代码省略.........

 main() {
  long loops;
  int rc;
  int size;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val;
  int dir;
  snd_pcm_uframes_t frames;
 unsigned char *buffer;
  int i=0;

/* socket setting */
   int sd;
   struct sockaddr_in s_addr;
   int n, n_send, status;

   int on = 1;


   sd = socket (AF_INET, SOCK_DGRAM, 0);

   bzero(&s_addr, sizeof(s_addr));
   s_addr.sin_family = AF_INET;
   s_addr.sin_addr.s_addr = inet_addr("192.168.42.255");
   s_addr.sin_port = htons(2007);

   if((status = setsockopt(sd, SOL_SOCKET, SO_BROADCAST, &on, sizeof(on))) != 0 )
   {
      printf("setsockopt error\n");
      exit(-1);
   }


  /* Open PCM device for recording (capture). */
  rc = snd_pcm_open(&handle, "plughw:1,0",
                    SND_PCM_STREAM_CAPTURE, 0);
  if (rc < 0) {
    fprintf(stderr,
            "unable to open pcm device: %s\n",
            snd_strerror(rc));
    exit(1);
  }

  /* Allocate a hardware parameters object. */
  snd_pcm_hw_params_alloca(&params);

  /* Fill it in with default values. */
  snd_pcm_hw_params_any(handle, params);

  /* Set the desired hardware parameters. */

  /* Interleaved mode */
  snd_pcm_hw_params_set_access(handle, params,
                      SND_PCM_ACCESS_RW_INTERLEAVED);

  /* Signed 16-bit little-endian format */
  snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_U8);

  /* Two channels (stereo) */
  snd_pcm_hw_params_set_channels(handle, params, 1);

  /* 44100 bits/second sampling rate (CD quality) */
  val = 44100;
  snd_pcm_hw_params_set_rate_near(handle, params,
                                  &val, &dir);

  /* Set period size to 32 frames. */
  frames =4800;
  snd_pcm_hw_params_set_period_size_near(handle,
                              params, &frames, &dir);

  /* Write the parameters to the driver */
  rc = snd_pcm_hw_params(handle, params);
  if (rc < 0) {
    fprintf(stderr,
            "unable to set hw parameters: %s\n",
            snd_strerror(rc));
    exit(1);
  }

  /* Use a buffer large enough to hold one period */
  snd_pcm_hw_params_get_period_size(params,
                                      &frames, &dir);
  size = frames *1; /* 2 bytes/sample, 2 channels */

  buffer = (unsigned char*) malloc(size);
	
  /* We want to loop for 5 seconds */
  snd_pcm_hw_params_get_period_time(params,
                                         &val, &dir);
  while (1) {
    rc = snd_pcm_readi(handle, buffer, frames);
    if (rc == -EPIPE) {
      /* EPIPE means overrun */
      fprintf(stderr, "overrun occurred\n");
      snd_pcm_prepare(handle);
    } else if (rc < 0) {
      fprintf(stderr,
              "error from read: %s\n",
//.........这里部分代码省略.........

int AudioAlsa::setHWParams( const ch_cnt_t _channels, snd_pcm_access_t _access )
{
	int err, dir;

	// choose all parameters
	if( ( err = snd_pcm_hw_params_any( m_handle, m_hwParams ) ) < 0 )
	{
		printf( "Broken configuration for playback: no configurations "
				"available: %s\n", snd_strerror( err ) );
		return err;
	}

	// set the interleaved read/write format
	if( ( err = snd_pcm_hw_params_set_access( m_handle, m_hwParams,
							_access ) ) < 0 )
	{
		printf( "Access type not available for playback: %s\n",
							snd_strerror( err ) );
		return err;
	}

	// set the sample format
	if( ( snd_pcm_hw_params_set_format( m_handle, m_hwParams,
						SND_PCM_FORMAT_S16_LE ) ) < 0 )
	{
		if( ( snd_pcm_hw_params_set_format( m_handle, m_hwParams,
						SND_PCM_FORMAT_S16_BE ) ) < 0 )
		{
			printf( "Neither little- nor big-endian available for "
					"playback: %s\n", snd_strerror( err ) );
			return err;
		}
		m_convertEndian = isLittleEndian();
	}
	else
	{
		m_convertEndian = !isLittleEndian();
	}

	// set the count of channels
	if( ( err = snd_pcm_hw_params_set_channels( m_handle, m_hwParams,
							_channels ) ) < 0 )
	{
		printf( "Channel count (%i) not available for playbacks: %s\n"
				"(Does your soundcard not support surround?)\n",
					_channels, snd_strerror( err ) );
		return err;
	}

	// set the sample rate
	if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
						sampleRate(), 0 ) ) < 0 )
	{
		if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
				mixer()->baseSampleRate(), 0 ) ) < 0 )
		{
			printf( "Could not set sample rate: %s\n",
							snd_strerror( err ) );
			return err;
		}
	}

	m_periodSize = mixer()->framesPerPeriod();
	m_bufferSize = m_periodSize * 8;
	dir = 0;
	err = snd_pcm_hw_params_set_period_size_near( m_handle, m_hwParams,
							&m_periodSize, &dir );
	if( err < 0 )
	{
		printf( "Unable to set period size %lu for playback: %s\n",
					m_periodSize, snd_strerror( err ) );
		return err;
	}
	dir = 0;
	err = snd_pcm_hw_params_get_period_size( m_hwParams, &m_periodSize,
									&dir );
	if( err < 0 )
	{
		printf( "Unable to get period size for playback: %s\n",
							snd_strerror( err ) );
	}

	dir = 0;
	err = snd_pcm_hw_params_set_buffer_size_near( m_handle, m_hwParams,
								&m_bufferSize );
	if( err < 0 )
	{
		printf( "Unable to set buffer size %lu for playback: %s\n",
					m_bufferSize, snd_strerror( err ) );
		return ( err );
	}
	err = snd_pcm_hw_params_get_buffer_size( m_hwParams, &m_bufferSize );

	if( 2 * m_periodSize > m_bufferSize )
	{
		printf( "buffer to small, could not use\n" );
		return ( err );
	}


//.........这里部分代码省略.........

static HRESULT SetFormat(IDsDriverBufferImpl *This, LPWAVEFORMATEX pwfx)
{
    snd_pcm_t *pcm = NULL;
    snd_pcm_hw_params_t *hw_params = This->hw_params;
    unsigned int buffer_time = 500000;
    snd_pcm_format_t format = -1;
    snd_pcm_uframes_t psize;
    DWORD rate = pwfx->nSamplesPerSec;
    int err=0;

    switch (pwfx->wBitsPerSample)
    {
        case  8: format = SND_PCM_FORMAT_U8; break;
        case 16: format = SND_PCM_FORMAT_S16_LE; break;
        case 24: format = SND_PCM_FORMAT_S24_3LE; break;
        case 32: format = SND_PCM_FORMAT_S32_LE; break;
        default: FIXME("Unsupported bpp: %d\n", pwfx->wBitsPerSample); return DSERR_GENERIC;
    }

    err = snd_pcm_open(&pcm, WOutDev[This->drv->wDevID].pcmname, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
    if (err < 0)
    {
        if (errno != EBUSY || !This->pcm)
        {
            WARN("Cannot open sound device: %s\n", snd_strerror(err));
            return DSERR_GENERIC;
        }
        snd_pcm_drop(This->pcm);
        snd_pcm_close(This->pcm);
        This->pcm = NULL;
        err = snd_pcm_open(&pcm, WOutDev[This->drv->wDevID].pcmname, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
        if (err < 0)
        {
            WARN("Cannot open sound device: %s\n", snd_strerror(err));
            return DSERR_BUFFERLOST;
        }
    }

    /* Set some defaults */
    snd_pcm_hw_params_any(pcm, hw_params);
    err = snd_pcm_hw_params_set_channels(pcm, hw_params, pwfx->nChannels);
    if (err < 0) { WARN("Could not set channels to %d\n", pwfx->nChannels); goto err; }

    err = snd_pcm_hw_params_set_format(pcm, hw_params, format);
    if (err < 0) { WARN("Could not set format to %d bpp\n", pwfx->wBitsPerSample); goto err; }

    /* Alsa's rate resampling is only used if the application specifically requests
     * a buffer at a certain frequency, else it is better to disable it due to unwanted
     * side effects, which may include: Less granular pointer, changing buffer sizes, etc
     */
#if SND_LIB_VERSION >= 0x010009
    snd_pcm_hw_params_set_rate_resample(pcm, hw_params, 0);
#endif

    err = snd_pcm_hw_params_set_rate_near(pcm, hw_params, &rate, NULL);
    if (err < 0) { rate = pwfx->nSamplesPerSec; WARN("Could not set rate\n"); goto err; }

    if (!ALSA_NearMatch(rate, pwfx->nSamplesPerSec))
    {
        WARN("Could not set sound rate to %d, but instead to %d\n", pwfx->nSamplesPerSec, rate);
        pwfx->nSamplesPerSec = rate;
        pwfx->nAvgBytesPerSec = rate * pwfx->nBlockAlign;
        /* Let DirectSound detect this */
    }

    snd_pcm_hw_params_set_periods_integer(pcm, hw_params);
    snd_pcm_hw_params_set_buffer_time_near(pcm, hw_params, &buffer_time, NULL);
    buffer_time = 10000;
    snd_pcm_hw_params_set_period_time_near(pcm, hw_params, &buffer_time, NULL);

    err = snd_pcm_hw_params_get_period_size(hw_params, &psize, NULL);
    buffer_time = 16;
    snd_pcm_hw_params_set_periods_near(pcm, hw_params, &buffer_time, NULL);

    if (!This->mmap)
    {
        HeapFree(GetProcessHeap(), 0, This->mmap_buffer);
        This->mmap_buffer = NULL;
    }

    err = snd_pcm_hw_params_set_access (pcm, hw_params, SND_PCM_ACCESS_MMAP_INTERLEAVED);
    if (err >= 0)
        This->mmap = 1;
    else
    {
        This->mmap = 0;
        err = snd_pcm_hw_params_set_access (pcm, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
    }

    err = snd_pcm_hw_params(pcm, hw_params);

    /* ALSA needs at least 3 buffers to work successfully */
    This->mmap_commitahead = 3 * psize;
    while (This->mmap_commitahead <= 512)
        This->mmap_commitahead += psize;

    if (This->pcm)
    {
        snd_pcm_drop(This->pcm);
        snd_pcm_close(This->pcm);
//.........这里部分代码省略.........

static RD_BOOL
alsa_set_format(snd_pcm_t * pcm, RD_WAVEFORMATEX * pwfx)
{
	snd_pcm_hw_params_t *hwparams = NULL;
	int err;
	unsigned int buffertime;
	short samplewidth;
	int audiochannels;
	unsigned int rate;

	samplewidth = pwfx->wBitsPerSample / 8;

	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
	{
		error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
		return False;
	}

	if ((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
	{
		error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
		return False;
	}

	if ((err = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
	{
		error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
		return False;
	}

	if (pwfx->wBitsPerSample == 16)
	{
		if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
		{
			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
			return False;
		}
	}
	else
	{
		if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S8)) < 0)
		{
			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
			return False;
		}
	}

#if 0
	if ((err = snd_pcm_hw_params_set_rate_resample(pcm, hwparams, 1)) < 0)
	{
		error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
		return False;
	}
#endif

	rate = pwfx->nSamplesPerSec;
	if ((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
	{
		error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
		return False;
	}

	audiochannels = pwfx->nChannels;
	if ((err = snd_pcm_hw_params_set_channels(pcm, hwparams, pwfx->nChannels)) < 0)
	{
		error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
		return False;
	}


	buffertime = 500000;	/* microseconds */
	if ((err = snd_pcm_hw_params_set_buffer_time_near(pcm, hwparams, &buffertime, 0)) < 0)
	{
		error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
		return False;
	}

	if ((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
	{
		error("snd_pcm_hw_params: %s\n", snd_strerror(err));
		return False;
	}

	snd_pcm_hw_params_free(hwparams);

	if ((err = snd_pcm_prepare(pcm)) < 0)
	{
		error("snd_pcm_prepare: %s\n", snd_strerror(err));
		return False;
	}

	reopened = True;

	return True;
}

static ALCboolean alsa_reset_playback(ALCdevice *device)
{
    alsa_data *data = (alsa_data*)device->ExtraData;
    snd_pcm_uframes_t periodSizeInFrames;
    unsigned int periodLen, bufferLen;
    snd_pcm_sw_params_t *sp = NULL;
    snd_pcm_hw_params_t *hp = NULL;
    snd_pcm_access_t access;
    snd_pcm_format_t format;
    unsigned int periods;
    unsigned int rate;
    const char *funcerr;
    int allowmmap;
    int err;

    format = -1;
    switch(device->FmtType)
    {
        case DevFmtByte:
            format = SND_PCM_FORMAT_S8;
            break;
        case DevFmtUByte:
            format = SND_PCM_FORMAT_U8;
            break;
        case DevFmtShort:
            format = SND_PCM_FORMAT_S16;
            break;
        case DevFmtUShort:
            format = SND_PCM_FORMAT_U16;
            break;
        case DevFmtInt:
            format = SND_PCM_FORMAT_S32;
            break;
        case DevFmtUInt:
            format = SND_PCM_FORMAT_U32;
            break;
        case DevFmtFloat:
            format = SND_PCM_FORMAT_FLOAT;
            break;
    }

    allowmmap = GetConfigValueBool("alsa", "mmap", 1);
    periods = device->NumUpdates;
    periodLen = (ALuint64)device->UpdateSize * 1000000 / device->Frequency;
    bufferLen = periodLen * periods;
    rate = device->Frequency;

    snd_pcm_hw_params_malloc(&hp);
#define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error
    CHECK(snd_pcm_hw_params_any(data->pcmHandle, hp));
    /* set interleaved access */
    if(!allowmmap || snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0)
    {
        if(periods > 2)
        {
            periods--;
            bufferLen = periodLen * periods;
        }
        CHECK(snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_RW_INTERLEAVED));
    }
    /* test and set format (implicitly sets sample bits) */
    if(snd_pcm_hw_params_test_format(data->pcmHandle, hp, format) < 0)
    {
        static const struct {
            snd_pcm_format_t format;
            enum DevFmtType fmttype;
        } formatlist[] = {
            { SND_PCM_FORMAT_FLOAT, DevFmtFloat  },
            { SND_PCM_FORMAT_S32,   DevFmtInt    },
            { SND_PCM_FORMAT_U32,   DevFmtUInt   },
            { SND_PCM_FORMAT_S16,   DevFmtShort  },
            { SND_PCM_FORMAT_U16,   DevFmtUShort },
            { SND_PCM_FORMAT_S8,    DevFmtByte   },
            { SND_PCM_FORMAT_U8,    DevFmtUByte  },
        };
        size_t k;

        for(k = 0;k < COUNTOF(formatlist);k++)
        {
            format = formatlist[k].format;
            if(snd_pcm_hw_params_test_format(data->pcmHandle, hp, format) >= 0)
            {
                device->FmtType = formatlist[k].fmttype;
                break;
            }
        }
    }
    CHECK(snd_pcm_hw_params_set_format(data->pcmHandle, hp, format));
    /* test and set channels (implicitly sets frame bits) */
    if(snd_pcm_hw_params_test_channels(data->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans)) < 0)
    {
        static const enum DevFmtChannels channellist[] = {
            DevFmtStereo,
            DevFmtQuad,
            DevFmtX51,
            DevFmtX71,
            DevFmtMono,
        };
        size_t k;

//.........这里部分代码省略.........

int main()
{
	int fp;
  unsigned int pcm, tmp, dir;
  int buff_size;
	
  long loops;
  int rc;
  int size;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val;
  
  snd_pcm_uframes_t frames;
  char *buff;
  int rate, channels, seconds;

  /* Open PCM device for recording (capture). */
  rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0);

  if (rc < 0) 
  {
    fprintf(stderr,"unable to open pcm device: %s\n",   snd_strerror(rc));
    exit(1);
  }

  /* Allocate a hardware parameters object. */
  snd_pcm_hw_params_alloca(&params);

  /* Fill it in with default values. */
  snd_pcm_hw_params_any(handle, params);

  /* Set the desired hardware parameters. */

  /* Interleaved mode */
  snd_pcm_hw_params_set_access(handle, params,  SND_PCM_ACCESS_RW_INTERLEAVED);

  /* Signed 16-bit little-endian format */
  snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_U8);

  /* Two channels (stereo) */
  snd_pcm_hw_params_set_channels(handle, params, 1);

  /* 44100 bits/second sampling rate (CD quality) */
  val = 8000;
  
  snd_pcm_hw_params_set_rate_near(handle, params,  &val, &dir);

  /* Set period size to 32 frames. */
  frames = 32;
  snd_pcm_hw_params_set_period_size_near(handle,params, &frames, &dir);

  /* Write the parameters to the driver */
  rc = snd_pcm_hw_params(handle, params);
  if (rc < 0) 
  {
    fprintf(stderr,"unable to set hw parameters: %s\n", s