view libao2/ao_alsa.c @ 18150:710d4bc5f8c9

Using channel count, samplerate and input bps values from the container instead of the decoder breaks some DTS samples where the container says the audio has 6 channels but the decoder gives 2. In this case take the number of channels from the decoder instead, the output will almost certainly be badly garbled anyway if the number of channels is wrong. patch by Uoti Urpala, uoti <<.>> urpala <<@>> pp1 <<.>> inet <<.>> fi

/*
  ao_alsa9/1.x - ALSA-0.9.x-1.x output plugin for MPlayer

  (C) Alex Beregszaszi
  
  modified for real alsa-0.9.0-support by Zsolt Barat <[e-mail]>
  additional AC3 passthrough support by Andy Lo A Foe <[e-mail]>  
  08/22/2002 iec958-init rewritten and merged with common init, zsolt
  04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
  04/25/2004 printfs converted to mp_msg, Zsolt.
  
  Any bugreports regarding to this driver are welcome.
*/

#include <errno.h>
#include <sys/time.h>
#include <stdlib.h>
#include <stdarg.h>
#include <math.h>
#include <string.h>

#include "config.h"
#include "subopt-helper.h"
#include "mixer.h"
#include "mp_msg.h"

#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API

#if HAVE_SYS_ASOUNDLIB_H
#include <sys/asoundlib.h>
#elif HAVE_ALSA_ASOUNDLIB_H
#include <alsa/asoundlib.h>
#else
#error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
#endif


#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"

static ao_info_t info = 
{
    "ALSA-0.9.x-1.x audio output",
    "alsa",
    "Alex Beregszaszi, Zsolt Barat <[e-mail]>",
    "under developement"
};

LIBAO_EXTERN(alsa)

static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
static snd_pcm_hw_params_t *alsa_hwparams;
static snd_pcm_sw_params_t *alsa_swparams;

/* 16 sets buffersize to 16 * chunksize is as default 1024
 * which seems to be good avarge for most situations 
 * so buffersize is 16384 frames by default */
static int alsa_fragcount = 16;
static snd_pcm_uframes_t chunk_size = 1024;

static size_t bytes_per_sample;

static int ao_noblock = 0;

static int open_mode;
static int alsa_can_pause = 0;

#define ALSA_DEVICE_SIZE 256

#undef BUFFERTIME
#define SET_CHUNKSIZE

static void alsa_error_handler(const char *file, int line, const char *function,
			       int err, const char *format, ...)
{
  char tmp[0xc00];
  va_list va;

  va_start(va, format);
  vsnprintf(tmp, sizeof tmp, format, va);
  va_end(va);
  tmp[sizeof tmp - 1] = '\0';

  if (err)
    mp_msg(MSGT_AO, MSGL_ERR, "alsa-lib: %s:%i:(%s) %s: %s\n",
	   file, line, function, tmp, snd_strerror(err));
  else
    mp_msg(MSGT_AO, MSGL_ERR, "alsa-lib: %s:%i:(%s) %s\n",
	   file, line, function, tmp);
}

/* to set/get/query special features/parameters */
static int control(int cmd, void *arg)
{
  switch(cmd) {
  case AOCONTROL_QUERY_FORMAT:
    return CONTROL_TRUE;
#ifndef WORDS_BIGENDIAN 
  case AOCONTROL_GET_VOLUME:
  case AOCONTROL_SET_VOLUME:
    {
      ao_control_vol_t *vol = (ao_control_vol_t *)arg;

      int err;
      snd_mixer_t *handle;
      snd_mixer_elem_t *elem;
      snd_mixer_selem_id_t *sid;

      static char *mix_name = "PCM";
      static char *card = "default";
      static int mix_index = 0;

      long pmin, pmax;
      long get_vol, set_vol;
      float f_multi;

      if(mixer_channel) {
	 char *test_mix_index;

	 mix_name = strdup(mixer_channel);
	 if ((test_mix_index = strchr(mix_name, ','))){
		*test_mix_index = 0;
		test_mix_index++;
		mix_index = strtol(test_mix_index, &test_mix_index, 0);

		if (*test_mix_index){
		  mp_msg(MSGT_AO,MSGL_ERR,
		    "alsa-control: invalid mixer index. Defaulting to 0\n");
		  mix_index = 0 ;
		}
	 }
      }
      if(mixer_device) card = mixer_device;

      if(ao_data.format == AF_FORMAT_AC3)
	return CONTROL_TRUE;

      //allocate simple id
      snd_mixer_selem_id_alloca(&sid);
	
      //sets simple-mixer index and name
      snd_mixer_selem_id_set_index(sid, mix_index);
      snd_mixer_selem_id_set_name(sid, mix_name);

      if (mixer_channel) {
	free(mix_name);
	mix_name = NULL;
      }

      if ((err = snd_mixer_open(&handle, 0)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer open error: %s\n", snd_strerror(err));
	return CONTROL_ERROR;
      }

      if ((err = snd_mixer_attach(handle, card)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer attach %s error: %s\n", 
	       card, snd_strerror(err));
	snd_mixer_close(handle);
	return CONTROL_ERROR;
      }

      if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer register error: %s\n", snd_strerror(err));
	snd_mixer_close(handle);
	return CONTROL_ERROR;
      }
      err = snd_mixer_load(handle);
      if (err < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer load error: %s\n", snd_strerror(err));
	snd_mixer_close(handle);
	return CONTROL_ERROR;
      }

      elem = snd_mixer_find_selem(handle, sid);
      if (!elem) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: unable to find simple control '%s',%i\n",
	       snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
	snd_mixer_close(handle);
	return CONTROL_ERROR;
	}

      snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
      f_multi = (100 / (float)(pmax - pmin));

      if (cmd == AOCONTROL_SET_VOLUME) {

	set_vol = vol->left / f_multi + pmin + 0.5;

	//setting channels
	if ((err = snd_mixer_selem_set_playback_volume(elem, 0, set_vol)) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: error setting left channel, %s\n", 
		 snd_strerror(err));
	  return CONTROL_ERROR;
	}
	mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);

	set_vol = vol->right / f_multi + pmin + 0.5;

	if ((err = snd_mixer_selem_set_playback_volume(elem, 1, set_vol)) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: error setting right channel, %s\n", 
		 snd_strerror(err));
	  return CONTROL_ERROR;
	}
	mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n", 
	       set_vol, pmin, pmax, f_multi);

	if (snd_mixer_selem_has_playback_switch(elem)) {
	  int lmute = (vol->left == 0.0);
	  int rmute = (vol->right == 0.0);
	  if (snd_mixer_selem_has_playback_switch_joined(elem)) {
	    lmute = rmute = lmute && rmute;
	  } else {
	    snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
	  }
	  snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
	}
      }
      else {
	snd_mixer_selem_get_playback_volume(elem, 0, &get_vol);
	vol->left = (get_vol - pmin) * f_multi;
	snd_mixer_selem_get_playback_volume(elem, 1, &get_vol);
	vol->right = (get_vol - pmin) * f_multi;

	mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
      }
      snd_mixer_close(handle);
      return CONTROL_OK;
    }
#endif
    
  } //end switch
  return(CONTROL_UNKNOWN);
}

static void parse_device (char *dest, const char *src, int len)
{
  char *tmp;
  memmove(dest, src, len);
  dest[len] = 0;
  while ((tmp = strrchr(dest, '.')))
    tmp[0] = ',';
  while ((tmp = strrchr(dest, '=')))
    tmp[0] = ':';
}

static void print_help (void)
{
  mp_msg (MSGT_AO, MSGL_FATAL,
           "\n-ao alsa commandline help:\n"
           "Example: mplayer -ao alsa:device=hw=0.3\n"
           "  sets first card fourth hardware device\n"
           "\nOptions:\n"
           "  noblock\n"
           "    Opens device in non-blocking mode\n"
           "  device=<device-name>\n"
           "    Sets device (change , to . and : to =)\n");
}

static int str_maxlen(strarg_t *str) {
  if (str->len > ALSA_DEVICE_SIZE)
    return 0;
  return 1;
}

/*
    open & setup audio device
    return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
    int err;
    int block;
    strarg_t device;
    snd_pcm_uframes_t bufsize;
    snd_pcm_uframes_t boundary;
    opt_t subopts[] = {
      {"block", OPT_ARG_BOOL, &block, NULL},
      {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen},
      {NULL}
    };

    char alsa_device[ALSA_DEVICE_SIZE + 1];
    // make sure alsa_device is null-terminated even when using strncpy etc.
    memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);

    mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
	channels, format);
    alsa_handler = NULL;
#if SND_LIB_VERSION >= 0x010005
    mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
#else
    mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
#endif

    snd_lib_error_set_handler(alsa_error_handler);
    
    ao_data.samplerate = rate_hz;
    ao_data.format = format;
    ao_data.channels = channels;

    switch (format)
      {
      case AF_FORMAT_S8:
	alsa_format = SND_PCM_FORMAT_S8;
	break;
      case AF_FORMAT_U8:
	alsa_format = SND_PCM_FORMAT_U8;
	break;
      case AF_FORMAT_U16_LE:
	alsa_format = SND_PCM_FORMAT_U16_LE;
	break;
      case AF_FORMAT_U16_BE:
	alsa_format = SND_PCM_FORMAT_U16_BE;
	break;
#ifndef WORDS_BIGENDIAN
      case AF_FORMAT_AC3:
#endif
      case AF_FORMAT_S16_LE:
	alsa_format = SND_PCM_FORMAT_S16_LE;
	break;
#ifdef WORDS_BIGENDIAN
      case AF_FORMAT_AC3:
#endif
      case AF_FORMAT_S16_BE:
	alsa_format = SND_PCM_FORMAT_S16_BE;
	break;
      case AF_FORMAT_U32_LE:
	alsa_format = SND_PCM_FORMAT_U32_LE;
	break;
      case AF_FORMAT_U32_BE:
	alsa_format = SND_PCM_FORMAT_U32_BE;
	break;
      case AF_FORMAT_S32_LE:
	alsa_format = SND_PCM_FORMAT_S32_LE;
	break;
      case AF_FORMAT_S32_BE:
	alsa_format = SND_PCM_FORMAT_S32_BE;
	break;
      case AF_FORMAT_FLOAT_LE:
	alsa_format = SND_PCM_FORMAT_FLOAT_LE;
	break;
      case AF_FORMAT_FLOAT_BE:
	alsa_format = SND_PCM_FORMAT_FLOAT_BE;
	break;
      case AF_FORMAT_MU_LAW:
	alsa_format = SND_PCM_FORMAT_MU_LAW;
	break;
      case AF_FORMAT_A_LAW:
	alsa_format = SND_PCM_FORMAT_A_LAW;
	break;

      default:
	alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
	break;
      }
    
    //subdevice parsing
    // set defaults
    block = 1;
    /* switch for spdif
     * sets opening sequence for SPDIF
     * sets also the playback and other switches 'on the fly'
     * while opening the abstract alias for the spdif subdevice
     * 'iec958'
     */
    if (format == AF_FORMAT_AC3) {
      unsigned char s[4];

	s[0] = IEC958_AES0_NONAUDIO | 
	  IEC958_AES0_CON_EMPHASIS_NONE;
	s[1] = IEC958_AES1_CON_ORIGINAL | 
	  IEC958_AES1_CON_PCM_CODER;
	s[2] = 0;
	s[3] = IEC958_AES3_CON_FS_48000;

	snprintf(alsa_device, ALSA_DEVICE_SIZE,
		"iec958:{CARD 0 AES0 0x%02x AES1 0x%02x AES2 0x%02x AES3 0x%02x}", 
 		s[0], s[1], s[2], s[3]);
	device.str = alsa_device;

	mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
    }
  else
        /* in any case for multichannel playback we should select
         * appropriate device
         */
        switch (channels) {
	case 1:
	case 2:
	  device.str = "default";
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
	  break;
	case 4:
	  if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
	    // hack - use the converter plugin
	    device.str = "plug:surround40";
	  else
	    device.str = "surround40";
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
	  break;
	case 6:
	  if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
	    device.str = "plug:surround51";
	  else
	    device.str = "surround51";
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
	  break;
	default:
	  device.str = "default";
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: %d channels are not supported\n",channels);
        }
    device.len = strlen(device.str);
    if (subopt_parse(ao_subdevice, subopts) != 0) {
        print_help();
        return 0;
    }
    ao_noblock = !block;
    parse_device(alsa_device, device.str, device.len);

    mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: using device %s\n", alsa_device);

    //setting modes for block or nonblock-mode
    if (ao_noblock) {
      open_mode = SND_PCM_NONBLOCK;
    }
    else {
      open_mode = 0;
    }

    //sets buff/chunksize if its set manually
    if (ao_data.buffersize) {
      switch (ao_data.buffersize)
	{
	case 1:
	  alsa_fragcount = 16;
	  chunk_size = 512;
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
	  break;
	case 2:
	  alsa_fragcount = 8;
	  chunk_size = 1024;
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
	  break;
	case 3:
	  alsa_fragcount = 32;
	  chunk_size = 512;
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
	  break;
	case 4:
	  alsa_fragcount = 16;
	  chunk_size = 1024;
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
	  break;
	default:
	  alsa_fragcount = 16;
	  chunk_size = 1024;
	  break;
	}
    }

    if (!alsa_handler) {
      //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
      if ((err = snd_pcm_open(&alsa_handler, alsa_device, SND_PCM_STREAM_PLAYBACK, open_mode)) < 0)
	{
	  if (err != -EBUSY && ao_noblock) {
	    mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: open in nonblock-mode failed, trying to open in block-mode\n");
	    if ((err = snd_pcm_open(&alsa_handler, alsa_device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
	      mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: playback open error: %s\n", snd_strerror(err));
	      return(0);
	    }
	  } else {
	    mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: playback open error: %s\n", snd_strerror(err));
	    return(0);
	  }
	}

      if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
         mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: error set block-mode %s\n", snd_strerror(err));
      } else {
	mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opend in blocking mode\n");
      }

      snd_pcm_hw_params_alloca(&alsa_hwparams);
      snd_pcm_sw_params_alloca(&alsa_swparams);

      // setting hw-parameters
      if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get initial parameters: %s\n",
		 snd_strerror(err));
	  return(0);
	}
    
      err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
					 SND_PCM_ACCESS_RW_INTERLEAVED);
      if (err < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set access type: %s\n", 
	       snd_strerror(err));
	return (0);
      }

      /* workaround for nonsupported formats
	 sets default format to S16_LE if the given formats aren't supported */
      if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
                                             alsa_format)) < 0)
      {
         mp_msg(MSGT_AO,MSGL_INFO,
		"alsa-init: format %s are not supported by hardware, trying default\n", af_fmt2str_short(format));
         alsa_format = SND_PCM_FORMAT_S16_LE;
         ao_data.format = AF_FORMAT_S16_LE;
      }

      if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
					      alsa_format)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set format: %s\n",
		 snd_strerror(err));
	  return(0);
	}

      if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
						     &ao_data.channels)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set channels: %s\n",
		 snd_strerror(err));
	  return(0);
	}

      /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
         prefer our own resampler */
#if SND_LIB_VERSION >= 0x010009
      if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
						     0)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to disable resampling: %s\n",
		 snd_strerror(err));
	  return(0);
	}
#endif

      if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, 
						 &ao_data.samplerate, NULL)) < 0) 
        {
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set samplerate-2: %s\n",
		 snd_strerror(err));
	  return(0);
        }

      bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
      bytes_per_sample *= ao_data.channels;
      ao_data.bps = ao_data.samplerate * bytes_per_sample;

#ifdef BUFFERTIME
      {
	int alsa_buffer_time = 500000; /* original 60 */
	int alsa_period_time;
	alsa_period_time = alsa_buffer_time/4;
	if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams, 
							  &alsa_buffer_time, NULL)) < 0)
	  {
	    mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set buffer time near: %s\n",
		   snd_strerror(err));
	    return(0);
	  } else
	    alsa_buffer_time = err;

	if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams, 
							  &alsa_period_time, NULL)) < 0)
	  /* original: alsa_buffer_time/ao_data.bps */
	  {
	    mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set period time: %s\n",
		   snd_strerror(err));
	  }
	mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: buffer_time: %d, period_time :%d\n",
	       alsa_buffer_time, err);
      } 
#endif//end SET_BUFFERTIME

#ifdef SET_CHUNKSIZE
      {
	//set chunksize
	if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams, 
							  &chunk_size, NULL)) < 0)
	  {
	    mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set periodsize(%ld): %s\n",
			    chunk_size, snd_strerror(err));
	  }
	else {
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size);
	}
	if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
						      &alsa_fragcount, NULL)) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set periods: %s\n", 
		 snd_strerror(err));
	}
	else {
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
	}
      }
#endif//end SET_CHUNKSIZE

      /* finally install hardware parameters */
      if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set hw-parameters: %s\n",
		 snd_strerror(err));
	}
      // end setting hw-params


      // gets buffersize for control
      if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get buffersize: %s\n", snd_strerror(err));
	}
      else {
	ao_data.buffersize = bufsize * bytes_per_sample;
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
      }

      if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get period size: %s\n", snd_strerror(err));
      } else {
	mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
      }
      ao_data.outburst = chunk_size * bytes_per_sample;

      /* setting software parameters */
      if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get sw-parameters: %s\n",
	       snd_strerror(err));
	return 0;
      }
#if SND_LIB_VERSION >= 0x000901
      if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get boundary: %s\n",
	       snd_strerror(err));
	return 0;
      }
#else
      boundary = 0x7fffffff;
#endif
      /* start playing when one period has been written */
      if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set start threshold: %s\n",
	       snd_strerror(err));
	return 0;
      }
      /* disable underrun reporting */
      if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set stop threshold: %s\n",
	       snd_strerror(err));
	return 0;
      }
#if SND_LIB_VERSION >= 0x000901
      /* play silence when there is an underrun */
      if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set silence size: %s\n",
	       snd_strerror(err));
	return 0;
      }
#endif
      if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set sw-parameters: %s\n",
	       snd_strerror(err));
	return 0;
      }
      /* end setting sw-params */

      mp_msg(MSGT_AO,MSGL_INFO,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
	     ao_data.samplerate, ao_data.channels, bytes_per_sample, ao_data.buffersize,
	     snd_pcm_format_description(alsa_format));

    } // end switch alsa_handler (spdif)
    alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
    return(1);
} // end init


/* close audio device */
static void uninit(int immed)
{

  if (alsa_handler) {
    int err;

    if (!immed)
      snd_pcm_drain(alsa_handler);

    if ((err = snd_pcm_close(alsa_handler)) < 0)
      {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: pcm close error: %s\n", snd_strerror(err));
	return;
      }
    else {
      alsa_handler = NULL;
      mp_msg(MSGT_AO,MSGL_INFO,"alsa-uninit: pcm closed\n");
    }
  }
  else {
    mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: no handler defined!\n");
  }
}

static void audio_pause(void)
{
    int err;

    if (alsa_can_pause) {
        if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
        {
            mp_msg(MSGT_AO,MSGL_ERR,"alsa-pause: pcm pause error: %s\n", snd_strerror(err));
            return;
        }
          mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
    } else {
        if ((err = snd_pcm_drop(alsa_handler)) < 0)
        {
            mp_msg(MSGT_AO,MSGL_ERR,"alsa-pause: pcm drop error: %s\n", snd_strerror(err));
            return;
        }
    }
}

static void audio_resume(void)
{
    int err;

    if (alsa_can_pause) {
        if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
        {
            mp_msg(MSGT_AO,MSGL_ERR,"alsa-resume: pcm resume error: %s\n", snd_strerror(err));
            return;
        }
          mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
    } else {
        if ((err = snd_pcm_prepare(alsa_handler)) < 0)
        {
           mp_msg(MSGT_AO,MSGL_ERR,"alsa-resume: pcm prepare error: %s\n", snd_strerror(err));
            return;
        }
    }
}

/* stop playing and empty buffers (for seeking/pause) */
static void reset(void)
{
    int err;

    if ((err = snd_pcm_drop(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-reset: pcm drop error: %s\n", snd_strerror(err));
	return;
    }
    if ((err = snd_pcm_prepare(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-reset: pcm prepare error: %s\n", snd_strerror(err));
	return;
    }
    return;
}

/*
    plays 'len' bytes of 'data'
    returns: number of bytes played
    modified last at 29.06.02 by jp
    thanxs for marius <[e-mail]> for giving us the light ;)
*/

static int play(void* data, int len, int flags)
{
  int num_frames = len / bytes_per_sample;
  snd_pcm_sframes_t res = 0;

  //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);

  if (!alsa_handler) {
    mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: device configuration error");
    return 0;
  }

  if (num_frames == 0)
    return 0;

  do {
    res = snd_pcm_writei(alsa_handler, data, num_frames);

      if (res == -EINTR) {
	/* nothing to do */
	res = 0;
      }
      else if (res == -ESTRPIPE) {	/* suspend */
	mp_msg(MSGT_AO,MSGL_INFO,"alsa-play: pcm in suspend mode. trying to resume\n");
	while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
	  sleep(1);
      }
      if (res < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: write error: %s\n", snd_strerror(res));
	mp_msg(MSGT_AO,MSGL_INFO,"alsa-play: trying to reset soundcard\n");
	if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: pcm prepare error: %s\n", snd_strerror(res));
	  return(0);
	  break;
	}
      }
  } while (res == 0);

  return res < 0 ? res : res * bytes_per_sample;
}

/* how many byes are free in the buffer */
static int get_space(void)
{
    snd_pcm_status_t *status;
    int ret;
    
    snd_pcm_status_alloca(&status);
    
    if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
    {
	mp_msg(MSGT_AO,MSGL_ERR,"alsa-space: cannot get pcm status: %s\n", snd_strerror(ret));
	return(0);
    }
    
    ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
    if (ret > MAX_OUTBURST)
	    ret = MAX_OUTBURST;
    return(ret);
}

/* delay in seconds between first and last sample in buffer */
static float get_delay(void)
{
  if (alsa_handler) {
    snd_pcm_sframes_t delay;
    
    if (snd_pcm_delay(alsa_handler, &delay) < 0)
      return 0;
    
    if (delay < 0) {
      /* underrun - move the application pointer forward to catch up */
#if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
      snd_pcm_forward(alsa_handler, -delay);
#endif
      delay = 0;
    }
    return (float)delay / (float)ao_data.samplerate;
  } else {
    return(0);
  }
}