alsa_a.c

MIDI解码程序(用VC编写)

  ret = snd_pcm_close (handle);
  if (ret < 0 && ret != -EINVAL) /* Maybe alsa-driver 0.5 has a bug */
    error_report (ret);
  handle = NULL;

  dpm.fd = -1;
}

static int output_data(char *buf, int32 nbytes)
{
  int n;
  snd_pcm_channel_status_t status;

  if (! handle)
    return -1;
  n = snd_pcm_write (handle, buf, nbytes);
  if (n <= 0)
    {
      memset (&status, 0, sizeof(status));
      status.channel = SND_PCM_CHANNEL_PLAYBACK;
      if (snd_pcm_channel_status(handle, &status) < 0)
	{
	  ctl->cmsg(CMSG_WARNING, VERB_DEBUG,
		    "%s: could not get channel status", alsa_device_name());
	  return -1;
	}
      if (status.status == SND_PCM_STATUS_UNDERRUN || n == -EPIPE)
	{
	  ctl->cmsg(CMSG_INFO, VERB_DEBUG,
		    "%s: underrun at %d", alsa_device_name(), status.scount);
	  output_counter += status.scount;
	  snd_pcm_channel_flush (handle, SND_PCM_CHANNEL_PLAYBACK);
	  snd_pcm_channel_prepare (handle, SND_PCM_CHANNEL_PLAYBACK);
	  n = snd_pcm_write (handle, buf, nbytes);
	}
      if (n <= 0)
	{
	  ctl->cmsg(CMSG_WARNING, VERB_DEBUG,
		    "%s: %s", alsa_device_name(),
		    (n < 0) ? snd_strerror(n) : "write error");
	  if (n != -EPIPE)  /* buffer underrun is ignored */
	    return -1;
	}
    }
  return 0;
}


static int acntl(int request, void *arg)
{
  int i;
  snd_pcm_channel_status_t pstatus;
  memset (&pstatus, 0, sizeof (pstatus));
  pstatus.channel = SND_PCM_CHANNEL_PLAYBACK;

  if (handle == NULL)
    return -1;

  switch (request)
    {
      case PM_REQ_GETFRAGSIZ:
	if (frag_size == 0)
	  return -1;
	*((int *)arg) = frag_size;
	return 0;

      case PM_REQ_GETQSIZ:
	if (total_bytes == -1)
	  return -1;
	*((int *)arg) = total_bytes;
	return 0;

      case PM_REQ_GETFILLABLE:
	if (total_bytes == -1)
	  return -1;
	if (snd_pcm_channel_status(handle, &pstatus) < 0)
	  return -1;
	i = pstatus.free >> sample_shift;
	*((int *)arg) = i;
	return 0;

      case PM_REQ_GETFILLED:
	if (total_bytes == -1)
	  return -1;
	if (snd_pcm_channel_status(handle, &pstatus) < 0)
	  return -1;
	i = pstatus.count >> sample_shift;
	*((int *)arg) = i;
	return 0;

      case PM_REQ_GETSAMPLES:
	if (total_bytes == -1)
	  return -1;
	if (snd_pcm_channel_status(handle, &pstatus) < 0)
	  return -1;
	i = (output_counter + pstatus.scount) >> sample_shift;
	*((int *)arg) = i;
	return 0;

      case PM_REQ_DISCARD:
	if (snd_pcm_playback_drain(handle) < 0)
	  return -1;
	if (snd_pcm_channel_prepare(handle, SND_PCM_CHANNEL_PLAYBACK) < 0)
	  return -1;
	output_counter = 0;
	return 0;

      case PM_REQ_FLUSH:
	if (snd_pcm_channel_flush(handle, SND_PCM_CHANNEL_PLAYBACK) < 0)
	  return -1;
	if (snd_pcm_channel_prepare(handle, SND_PCM_CHANNEL_PLAYBACK) < 0)
	  return -1;
	output_counter = 0;
	return 0;
    }
    return -1;
}

/* end ALSA API 0.5.x */

#elif ALSA_LIB < 5

/*================================================================
 * ALSA API version 0.4.x
 *================================================================*/

/*return value == 0 sucess
               == 1 warning
               == -1 fails
 */
static int set_playback_info (snd_pcm_t* handle__,
			      int32* encoding__, int32* rate__,
			      const int32 extra_param[5])
{
  int ret_val = 0;
  const int32 orig_encoding = *encoding__;
  const int32 orig_rate = *rate__;
  int tmp;
  snd_pcm_playback_info_t pinfo;
  snd_pcm_format_t pcm_format;
  struct snd_pcm_playback_params pparams;
  struct snd_pcm_playback_status pstatus;
  memset (&pcm_format, 0, sizeof (pcm_format));

  memset (&pinfo, 0, sizeof (pinfo));
  memset (&pparams, 0, sizeof (pparams));
  tmp = snd_pcm_playback_info (handle__, &pinfo);
  if (tmp < 0)
    {
      error_report (tmp);
      return -1;
    }

  /*check sample bit*/
  if ((pinfo.flags & SND_PCM_PINFO_8BITONLY) != 0)
    *encoding__ &= ~PE_16BIT; /*force 8bit samples*/
  if ((pinfo.flags & SND_PCM_PINFO_16BITONLY) != 0)
    *encoding__ |= PE_16BIT; /*force 16bit samples*/

  /*check rate*/
  if (pinfo.min_rate > *rate__)
    *rate__ = pinfo.min_rate;
  if (pinfo.max_rate < *rate__)
    *rate__ = pinfo.max_rate;
  pcm_format.rate = *rate__;

  /*check channels*/
  if ((*encoding__ & PE_MONO) != 0 && pinfo.min_channels > 1)
    *encoding__ &= ~PE_MONO;
  if ((*encoding__ & PE_MONO) == 0 && pinfo.max_channels < 2)
    *encoding__ |= PE_MONO;

  if ((*encoding__ & PE_MONO) != 0)
    pcm_format.channels = 1; /*mono*/
  else
    pcm_format.channels = 2; /*stereo*/

  /*check format*/
  if ((*encoding__ & PE_16BIT) != 0)
    { /*16bit*/
      if ((pinfo.formats & SND_PCM_FMT_S16_LE) != 0)
	{
	  pcm_format.format = SND_PCM_SFMT_S16_LE;
	  *encoding__ |= PE_SIGNED;
	}
      else
	{
	  ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
		    "%s doesn't support 16 bit sample width",
		    alsa_device_name());
	  return -1;
	}
    }
  else
    { /*8bit*/
      if ((pinfo.formats & SND_PCM_FMT_U8) != 0)
	{
	  pcm_format.format = SND_PCM_SFMT_U8;
	  *encoding__ &= ~PE_SIGNED;
	}
#if 0
      else if ((pinfo.formats & SND_PCM_FMT_S8) != 0)
	{
	  pcm_format.format = SND_PCM_SFMT_U16_LE;
	  *encoding__ |= PE_SIGNED;
	}
#endif
      else
	{
	  ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
		    "%s doesn't support 8 bit sample width",
		    alsa_device_name());
	  return -1;
	}
    }


  tmp = snd_pcm_playback_format (handle__, &pcm_format);
  if (tmp < 0)
    {
      error_report (tmp);
      return -1;
    }
  /*check result of snd_pcm_playback_format*/
  if ((*encoding__ & PE_16BIT) != (orig_encoding & PE_16BIT ))
    {
      ctl->cmsg (CMSG_WARNING, VERB_VERBOSE,
		 "Sample width adjusted to %d bits",
		 ((*encoding__ & PE_16BIT) != 0)? 16:8);
      ret_val = 1;
    }
  if (((pcm_format.channels == 1)? PE_MONO:0) != (orig_encoding & PE_MONO))
    {
      ctl->cmsg(CMSG_WARNING, VERB_VERBOSE, "Sound adjusted to %sphonic",
		((*encoding__ & PE_MONO) != 0)? "mono" : "stereo");
      ret_val = 1;
    }

  sample_shift = 0;
  if (!(dpm.encoding & PE_MONO))
    sample_shift++;
  if (dpm.encoding & PE_16BIT)
    sample_shift++;

  /* Set buffer fragment size (in extra_param[1]) */
  if (extra_param[1] != 0)
    tmp = extra_param[1];
  else
    tmp = audio_buffer_size << sample_shift;

  /* Set buffer fragments (in extra_param[0]) */
  pparams.fragment_size  = tmp;
  pparams.fragments_max  = (extra_param[0] == 0) ? -1 : extra_param[0];
  pparams.fragments_room = 1;
  tmp = snd_pcm_playback_params (handle__, &pparams);
  if (tmp < 0)
    {
      ctl->cmsg(CMSG_WARNING, VERB_NORMAL,
		"%s doesn't support buffer fragments"
		":request size=%d, max=%d, room=%d\n",
		alsa_device_name(),
		pparams.fragment_size,
		pparams.fragments_max,
		pparams.fragments_room);
      ret_val = 1;
    }

  if (snd_pcm_playback_status(handle__, &pstatus) == 0)
    {
      if (pstatus.rate != orig_rate)
	{
	  ctl->cmsg(CMSG_WARNING, VERB_VERBOSE,
		    "Output rate adjusted to %d Hz (requested %d Hz)",
		    pstatus.rate, orig_rate);
	  dpm.rate = pstatus.rate;
	  ret_val = 1;
	}
      frag_size = pstatus.fragment_size;
      total_bytes = pstatus.count;
    }
  else
    {
      frag_size = 0;
      total_bytes = -1; /* snd_pcm_playback_status fails */
    }

  return ret_val;
}

static int open_output(void)
{
  int tmp, warnings=0;
  int ret;

  tmp = check_sound_cards (&card, &device, dpm.extra_param);
  if (tmp < 0)
    return -1;

  /* Open the audio device */
  ret = snd_pcm_open (&handle, card, device, SND_PCM_OPEN_PLAYBACK);
  if (ret < 0)
    {
      ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: %s",
		alsa_device_name(), snd_strerror (ret));
      return -1;
    }

  /* They can't mean these */
  dpm.encoding &= ~(PE_ULAW|PE_ALAW|PE_BYTESWAP);
  warnings = set_playback_info (handle, &dpm.encoding, &dpm.rate,
				dpm.extra_param);
  if (warnings < 0)
    {
      close_output ();
      return -1;
    }

  dpm.fd = snd_pcm_file_descriptor (handle);
  output_counter = 0;
  return warnings;
}

static void close_output(void)
{
  int ret;

  if (handle == NULL)
    return;

  ret = snd_pcm_close (handle);
  if (ret < 0)
    error_report (ret);
  handle = NULL;

  dpm.fd = -1;
}

static int output_data(char *buf, int32 nbytes)
{
  int n;

  if (! handle)
    return -1;
  while (nbytes > 0)
    {
      n = snd_pcm_write (handle, buf, nbytes);

      if (n < 0)
        {
	  ctl->cmsg(CMSG_WARNING, VERB_DEBUG,
		    "%s: %s", alsa_device_name(), snd_strerror(n));
	  if (n == -EWOULDBLOCK)
	    continue;
	  return -1;
	}
      buf += n;
      nbytes -= n;
      output_counter += n;
    }

  return 0;
}

static int acntl(int request, void *arg)
{
  int i;
  struct snd_pcm_playback_status pstatus;

  if (handle == NULL)
    return -1;

  switch (request)
    {
      case PM_REQ_GETFRAGSIZ:
	if (frag_size == 0)
	  return -1;
	*((int *)arg) = frag_size;
	return 0;

      case PM_REQ_GETQSIZ:
	if (total_bytes == -1)
	  return -1;
	*((int *)arg) = total_bytes;
	return 0;

      case PM_REQ_GETFILLABLE:
	if (total_bytes == -1)
	  return -1;
	if (snd_pcm_playback_status(handle, &pstatus) < 0)
	  return -1;
	i = pstatus.count >> sample_shift;
	*((int *)arg) = i;
	return 0;

      case PM_REQ_GETFILLED:
	if (total_bytes == -1)
	  return -1;
	if (snd_pcm_playback_status(handle, &pstatus) < 0)
	  return -1;
	i = pstatus.queue >> sample_shift;
	*((int *)arg) = i;
	return 0;

      case PM_REQ_GETSAMPLES:
	if (total_bytes == -1)
	  return -1;
	if (snd_pcm_playback_status(handle, &pstatus) < 0)
	  return -1;
	i = output_counter - pstatus.queue;
	i >>= sample_shift;
	*((int *)arg) = i;
	return 0;

      case PM_REQ_DISCARD:
	if (snd_pcm_drain_playback (handle) < 0)
	  return -1; /* error */
	output_counter = 0;
	return 0;

      case PM_REQ_FLUSH:
	if (snd_pcm_flush_playback(handle) < 0)
	  return -1; /* error */
	output_counter = 0;
	return 0;
    }
    return -1;
}

/* end ALSA API 0.4.x */

#endif