src/audio_out/audio_alsa_out.c

/*
 * Copyright (C) 2000-2020 the xine project
 *
 * This file is part of xine, a free video player.
 *
 * xine is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * xine is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110, USA
 *
 * Credits go
 * - for the SPDIF A/52 sync part
 * - frame size calculation added (16-08-2001)
 * (c) 2001 Andy Lo A Foe <andy@alsaplayer.org>
 * for initial ALSA 0.9.x support.
 *     adding MONO/STEREO/4CHANNEL/5CHANNEL/5.1CHANNEL analogue support.
 * (c) 2001 James Courtier-Dutton <James@superbug.demon.co.uk>
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include <stdio.h>
#include <stdarg.h>
#include <errno.h>
#include <string.h>
#include <unistd.h>
#include <stdlib.h>
#include <fcntl.h>
#include <math.h>
#ifdef HAVE_ALLOCA_H
#include <alloca.h>
#endif

#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>

#include <sys/ioctl.h>
#include <inttypes.h>
#include <pthread.h>

#include <xine/xine_internal.h>
#include <xine/xineutils.h>
#include <xine/compat.h>
#include <xine/audio_out.h>

#include "speakers.h"

/*
#define ALSA_LOG
#define ALSA_LOG_BUFFERS
*/
/*
#define LOG_DEBUG
*/

#define AO_OUT_ALSA_IFACE_VERSION 9

#define BUFFER_TIME               1000*1000
#define GAP_TOLERANCE             5000

#define MIXER_MASK_LEFT           0x0001
#define MIXER_MASK_RIGHT          0x0002
#define MIXER_MASK_MUTE           0x0004
#define MIXER_MASK_STEREO         0x0008
#define MIXER_HAS_MUTE_SWITCH     0x0010

typedef struct {
  audio_driver_class_t driver_class;

  xine_t          *xine;
} alsa_class_t;

typedef struct alsa_driver_s {

  ao_driver_t        ao_driver;

  alsa_class_t      *class;

  snd_pcm_t         *audio_fd;
  int                open_mode;
  int		     has_pause_resume;
  int		     is_paused;

  int32_t            output_sample_rate, input_sample_rate;
  double             sample_rate_factor;
  uint32_t           num_channels;
  uint32_t           bits_per_sample;
  uint32_t           bytes_per_frame;
  uint32_t           bytes_in_buffer;      /* number of bytes writen to audio hardware   */
  snd_pcm_uframes_t  buffer_size;
  int32_t            mmap;
  uint32_t           supported_channels;   /* field of 1 << num_channels */
  uint32_t           capabilities;

  struct _alsa_dev_info_s {
    struct alsa_driver_s *this;
    const char           *type;
    const char           *config_key;
    char                 *name;
    uint32_t              supported_channels;   /* field of 1 << num_channels */
    uint32_t              capabilities;
  }                    devs[5];

  const char          *bits_names[4]; /* 8, 16, 24, 32 */

  struct {
    pthread_t          thread;
    int                thread_created;
    pthread_mutex_t    mutex;
    char              *name;
    snd_mixer_t       *handle;
    snd_mixer_elem_t  *elem;
    long               min;
    long               max;
    long               left_vol;
    long               right_vol;
    int                mute;
    int                running;
  } mixer;

  /* avoid *_alloca () */
  snd_pcm_hw_params_t   *hw_params;
  snd_pcm_sw_params_t   *sw_params;
  snd_pcm_access_mask_t *ac_mask;
  snd_ctl_card_info_t   *card_info;
  snd_pcm_status_t      *pcm_status;
} alsa_driver_t;

static snd_output_t *jcd_out;

/*
 * Get and convert volume to percent value
 */
static int ao_alsa_get_percent_from_volume(long val, long min, long max) {
  int range = max - min;
  return (range == 0) ? 0 : ((val - min) * 100.0 / range + .5);
}

/* Stolen from alsa-lib */
static int my_snd_mixer_wait(snd_mixer_t *mixer, int timeout) {
  struct pollfd  spfds[16];
  struct pollfd *pfds = spfds;
  int            err, count;
  void          *freeme = NULL;

  count = snd_mixer_poll_descriptors(mixer, pfds, sizeof(spfds) / sizeof(spfds[0]));

  if (count < 0)
    return count;

  if ((unsigned int) count > sizeof(spfds) / sizeof(spfds[0])) {
    freeme = pfds = calloc(count, sizeof(*pfds));

    if (!pfds)
      return -ENOMEM;

    err = snd_mixer_poll_descriptors(mixer, pfds, (unsigned int) count);
    // XXX is this assert correct ? if triggered, it would be alsa bug, not xine bug ?
    assert(err == count);
    if (err < 0) {
      free(freeme);
      return err;
    }
  }

  err = poll(pfds, (unsigned int) count, timeout);

  free(freeme);

  if (err < 0)
    return -errno;

  return err;
}

/*
 * Wait (non blocking) till a mixer event happen
 */
static void *ao_alsa_handle_event_thread(void *data) {
  alsa_driver_t  *this = (alsa_driver_t *) data;

  do {

    if(my_snd_mixer_wait(this->mixer.handle, 333) > 0) {
      int err, mute = 0, swl = 0, swr = 0;
      long right_vol, left_vol;
      int old_mute;

      pthread_mutex_lock(&this->mixer.mutex);

      old_mute = (this->mixer.mute & MIXER_MASK_MUTE) ? 1 : 0;

      if((err = snd_mixer_handle_events(this->mixer.handle)) < 0) {
	xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: snd_mixer_handle_events(): %s\n",  snd_strerror(err));
	pthread_mutex_unlock(&this->mixer.mutex);
	continue;
      }

      if((err = snd_mixer_selem_get_playback_volume(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT, &left_vol)) < 0) {
	xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: snd_mixer_selem_get_playback_volume(): %s\n",  snd_strerror(err));
	pthread_mutex_unlock(&this->mixer.mutex);
	continue;
      }

      if((err = snd_mixer_selem_get_playback_volume(this->mixer.elem, SND_MIXER_SCHN_FRONT_RIGHT, &right_vol)) < 0) {
	xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: snd_mixer_selem_get_playback_volume(): %s\n",  snd_strerror(err));
	pthread_mutex_unlock(&this->mixer.mutex);
	continue;
      }

      if(this->mixer.mute & MIXER_HAS_MUTE_SWITCH) {

	if(this->mixer.mute & MIXER_MASK_STEREO) {
	  snd_mixer_selem_get_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT, &swl);
	  mute = (swl) ? 0 : 1;
	}
	else {

	  if (this->mixer.mute & MIXER_MASK_LEFT)
	    snd_mixer_selem_get_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT, &swl);

	  if ((SND_MIXER_SCHN_FRONT_RIGHT != SND_MIXER_SCHN_UNKNOWN) && (this->mixer.mute & MIXER_MASK_RIGHT))
	    snd_mixer_selem_get_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_RIGHT, &swr);

	  mute = (swl || swr) ? 0 : 1;
	}
      }

      if((this->mixer.right_vol != right_vol) || (this->mixer.left_vol != left_vol) || (old_mute != mute)) {
	xine_event_t              event;
	xine_audio_level_data_t   data;
	xine_stream_t            *stream;
	xine_list_iterator_t      ite;

	this->mixer.right_vol = right_vol;
	this->mixer.left_vol  = left_vol;
	if(mute)
	  this->mixer.mute |= MIXER_MASK_MUTE;
	else
	  this->mixer.mute &= ~MIXER_MASK_MUTE;

	data.right = ao_alsa_get_percent_from_volume(this->mixer.right_vol,
						     this->mixer.min, this->mixer.max);
	data.left  = ao_alsa_get_percent_from_volume(this->mixer.left_vol,
						     this->mixer.min, this->mixer.max);
	data.mute  = (this->mixer.mute & MIXER_MASK_MUTE) ? 1 : 0;

	event.type        = XINE_EVENT_AUDIO_LEVEL;
	event.data        = &data;
	event.data_length = sizeof(data);

	pthread_mutex_lock(&this->class->xine->streams_lock);
        ite = NULL;
        while ((stream = xine_list_next_value (this->class->xine->streams, &ite))) {
	  event.stream = stream;
	  xine_event_send(stream, &event);
	}
	pthread_mutex_unlock(&this->class->xine->streams_lock);
      }

      pthread_mutex_unlock(&this->mixer.mutex);
    }

  } while(this->mixer.running);

  pthread_exit(NULL);
}

/*
 * Convert percent value to volume and set
 */
static long ao_alsa_get_volume_from_percent(int val, long min, long max) {
  int range = max - min;
  return (range == 0) ? min : (val * range / 100.0 + min + .5);
}

/*
 * Error callback, we need to control this,
 * error message should be printed only in DEBUG mode.
 * XINE_FORMAT_PRINTF(5, 6) is true but useless here,
 * as alsa delivers "fmt" at runtime only.
 */
static void error_callback(const char *file, int line,
			   const char *function, int err, const char *fmt, ...) {
#ifdef DEBUG
  va_list   args;
  char     *buf = NULL;
  int       result;

  va_start(args, fmt);
  result = vasprintf(&buf, fmt, args);
  va_end(args);

  if (result >= 0) {
    printf("%s: %s() %s.\n", file, function, buf);
    free(buf);
  }
#else
  (void)file;
  (void)function;
  (void)fmt;
#endif
  (void)err;
  (void)line;
}

/*
 * open the audio device for writing to
 */
static int ao_alsa_open(ao_driver_t *this_gen, uint32_t bits, uint32_t rate, int mode) {
  alsa_driver_t        *this = (alsa_driver_t *) this_gen;
  char                 *pcm_device;
  int                   err;

  err = snd_output_stdio_attach(&jcd_out, stdout, 0);

  switch (mode) {
  case AO_CAP_MODE_MONO:
    this->num_channels = 1;
    pcm_device = this->devs[0].name;
    break;
  case AO_CAP_MODE_STEREO:
    this->num_channels = 2;
    pcm_device = this->devs[1].name;
    break;
  case AO_CAP_MODE_4CHANNEL:
    this->num_channels = 4;
    pcm_device = this->devs[2].name;
    break;
  case AO_CAP_MODE_4_1CHANNEL:
  case AO_CAP_MODE_5CHANNEL:
  case AO_CAP_MODE_5_1CHANNEL:
    this->num_channels = 6;
    pcm_device = this->devs[3].name;
    break;
  case AO_CAP_MODE_A52:
  case AO_CAP_MODE_AC5:
    this->num_channels = 2;
    pcm_device = this->devs[4].name;
    break;
  default:
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
	     "audio_alsa_out: ALSA Driver does not support the requested mode: 0x%X\n",mode);
    return 0;
  }

#ifdef ALSA_LOG
  printf("audio_alsa_out: Audio Device name = %s\n",pcm_device);
  printf("audio_alsa_out: Number of channels = %d\n",this->num_channels);
#endif

  if (this->audio_fd) {
    xine_log (this->class->xine, XINE_LOG_MSG, _("audio_alsa_out:Already open...WHY!"));
    snd_pcm_close (this->audio_fd);
    this->audio_fd = NULL;
  }

  this->open_mode              = mode;
  this->input_sample_rate      = rate;
  this->bits_per_sample        = bits;
  this->bytes_in_buffer        = 0;

  /*
   * open audio device
   * When switching to surround, dmix blocks the device some time, so we just keep trying for 0.8sec.
   */
  {
    struct timeval start_time, end_time;
    gettimeofday (&start_time, NULL);
    while (1) {
      err = snd_pcm_open (&this->audio_fd, pcm_device, SND_PCM_STREAM_PLAYBACK, /* NONBLOCK */ 1);
      if (err != -EBUSY)
        break;
      gettimeofday (&end_time, NULL);
      if ((end_time.tv_sec - start_time.tv_sec) * 1000000 + end_time.tv_usec - start_time.tv_usec > 800000)
        break;
      usleep (10000);
    }
  }
  if (err < 0) {
    xprintf (this->class->xine, XINE_VERBOSITY_LOG,
      _("audio_alsa_out: snd_pcm_open() of %s failed: %s\n"), pcm_device, snd_strerror (err));
    if (err == -EBUSY) {
      xprintf (this->class->xine, XINE_VERBOSITY_LOG,
        _("audio_alsa_out: >>> check if another program already uses PCM <<<\n"));
    }
    return 0;
  }
  /* printf ("audio_alsa_out: snd_pcm_open() opened %s\n", pcm_device); */
  /* We wanted non blocking open but now put it back to normal */
  //snd_pcm_nonblock(this->audio_fd, 0);
  snd_pcm_nonblock(this->audio_fd, 1);

  do {
    /* configure audio device */
    err = snd_pcm_hw_params_any (this->audio_fd, this->hw_params);
    if (err < 0) {
      xprintf (this->class->xine, XINE_VERBOSITY_LOG,
        _("audio_alsa_out: broken configuration for this PCM: no configurations available: %s\n"),
        snd_strerror (err));
      break;
    }

    /* set interleaved access */
    do {
      if (this->mmap != 0) {
        snd_pcm_access_mask_none (this->ac_mask);
        snd_pcm_access_mask_set (this->ac_mask, SND_PCM_ACCESS_MMAP_INTERLEAVED);
        snd_pcm_access_mask_set (this->ac_mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
        snd_pcm_access_mask_set (this->ac_mask, SND_PCM_ACCESS_MMAP_COMPLEX);
        err = snd_pcm_hw_params_set_access_mask (this->audio_fd, this->hw_params, this->ac_mask);
        if (err >= 0)
          break;
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: mmap not available, falling back to compatiblity mode\n");
        this->mmap = 0;
      }
      err = snd_pcm_hw_params_set_access (this->audio_fd, this->hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
    } while (0);
    if (err < 0) {
      xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
        "audio_alsa_out: access type not available: %s\n", snd_strerror (err));
      break;
    }

    /* set the sample format ([SU]{8,16,24,FLOAT}) */
    /* ALSA automatically appends _LE or _BE depending on the CPU */
    if ((bits < 8) || (bits > 32)) {
      xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
        "audio_alsa_out: pcm format bits=%d unknown, trying 16.\n", (int)bits);
      bits = 16;
    }
    {
      static const snd_pcm_format_t fmts[5] = {
        0,
        SND_PCM_FORMAT_U8,
        SND_PCM_FORMAT_S16,
#ifdef WORDS_BIGENDIAN
        SND_PCM_FORMAT_S24_3BE, /* 24 bit samples taking 3 bytes. */
#else
        SND_PCM_FORMAT_S24_3LE,
#endif
        SND_PCM_FORMAT_FLOAT
      };
      snd_pcm_format_t format = fmts[bits >> 3];
      err = snd_pcm_hw_params_set_format (this->audio_fd, this->hw_params, format);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: sample format not available: %s\n", snd_strerror (err));
        break;
      }
    }

    /* set the number of channels */
    err = snd_pcm_hw_params_set_channels (this->audio_fd, this->hw_params, this->num_channels);
    if (err < 0) {
      xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
        "audio_alsa_out: Cannot set number of channels to %d (err=%d:%s)\n",
        this->num_channels, err, snd_strerror (err));
      break;
    }

    {
      snd_pcm_uframes_t period_size, period_size_min, period_size_max;
      snd_pcm_uframes_t buffer_size_min, buffer_size_max;
      uint32_t buffer_time = BUFFER_TIME;
      snd_pcm_uframes_t buffer_time_to_size;
      int dir;
#if 0
      uint32_t              periods;
#endif

#if 0
      /* Restrict a configuration space to contain only real hardware rates */
      err = snd_pcm_hw_params_set_rate_resample (this->audio_fd, this->hw_params, 0);
#endif
      /* set the stream rate [Hz] */
      dir = 0;
      err = snd_pcm_hw_params_set_rate_near (this->audio_fd, this->hw_params, &rate, &dir);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: rate not available: %s\n", snd_strerror (err));
        break;
      }
      this->output_sample_rate = (uint32_t)rate;
      if (this->input_sample_rate != this->output_sample_rate) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: audio rate : %d requested, %d provided by device/sec\n",
            this->input_sample_rate, this->output_sample_rate);
      }

      buffer_time_to_size = ((uint64_t)buffer_time * rate) / 1000000;
      snd_pcm_hw_params_get_buffer_size_min (this->hw_params, &buffer_size_min);
      snd_pcm_hw_params_get_buffer_size_max (this->hw_params, &buffer_size_max);
      dir = 0;
      snd_pcm_hw_params_get_period_size_min (this->hw_params, &period_size_min, &dir);
      dir = 0;
      snd_pcm_hw_params_get_period_size_max (this->hw_params, &period_size_max, &dir);
#ifdef ALSA_LOG_BUFFERS
      printf ("Buffer size range from %lu to %lu\n", buffer_size_min, buffer_size_max);
      printf ("Period size range from %lu to %lu\n", period_size_min, period_size_max);
      printf ("Buffer time size %lu\n", buffer_time_to_size);
#endif
      this->buffer_size = buffer_time_to_size;
      if (buffer_size_max < this->buffer_size)
        this->buffer_size = buffer_size_max;
      if (buffer_size_min > this->buffer_size)
        this->buffer_size = buffer_size_min;
      period_size = this->buffer_size / 8;
      this->buffer_size = period_size * 8;
#ifdef ALSA_LOG_BUFFERS
      printf ("To choose buffer_size = %ld\n", this->buffer_size);
      printf ("To choose period_size = %ld\n", period_size);
#endif

#if 0
      /* Set period to buffer size ratios at 8 periods to 1 buffer */
      dir = -1;
      periods = 8;
      err = snd_pcm_hw_params_set_periods_near (this->audio_fd, this->hw_params, &periods, &dir);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: unable to set any periods: %s\n", snd_strerror (err));
        break;
      }
      /* set the ring-buffer time [us] (large enough for x us|y samples ...) */
      dir = 0;
      err = snd_pcm_hw_params_set_buffer_time_near (this->audio_fd, this->hw_params, &buffer_time, &dir);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: buffer time not available: %s\n", snd_strerror (err));
        break;
      }
#endif

#if 1
      /* set the period time [us] (interrupt every x us|y samples ...) */
      dir = 0;
      err = snd_pcm_hw_params_set_period_size_near (this->audio_fd, this->hw_params, &period_size, &dir);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: period time not available: %s\n", snd_strerror (err));
        break;
      }
#endif
      dir = 0;
      snd_pcm_hw_params_get_period_size (this->hw_params, &period_size, &dir);

      dir = 0;
      err = snd_pcm_hw_params_set_buffer_size_near (this->audio_fd, this->hw_params, &this->buffer_size);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: buffer time not available: %s\n", snd_strerror (err));
        break;
      }
      snd_pcm_hw_params_get_buffer_size (this->hw_params, &(this->buffer_size));
#ifdef ALSA_LOG_BUFFERS
      printf ("was set period_size = %ld\n", period_size);
      printf ("was set buffer_size = %ld\n", this->buffer_size);
#endif
      if (2 * period_size > this->buffer_size) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: buffer too small, could not use\n");
        break;
      }

      /* write the parameters to device */
      err = snd_pcm_hw_params (this->audio_fd, this->hw_params);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: pcm hw_params failed: %s\n", snd_strerror (err));
        break;
      }

      /* Check for pause/resume support */
      this->has_pause_resume = (snd_pcm_hw_params_can_pause (this->hw_params)
                             && snd_pcm_hw_params_can_resume (this->hw_params));
      xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
        "audio_alsa_out:open pause_resume=%d\n", this->has_pause_resume);
      if (this->has_pause_resume)
        this->capabilities &= ~AO_CAP_NO_UNPAUSE;
      else
        this->capabilities |= AO_CAP_NO_UNPAUSE;

      this->sample_rate_factor = (double) this->output_sample_rate / (double) this->input_sample_rate;
      this->bytes_per_frame = snd_pcm_frames_to_bytes (this->audio_fd, 1);

      /* audio buffer size handling */
      /* Copy current parameters into swparams */
      err = snd_pcm_sw_params_current (this->audio_fd, this->sw_params);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: Unable to determine current swparams: %s\n", snd_strerror (err));
        break;
      }

#if defined(SND_LIB_VERSION) && SND_LIB_VERSION >= 0x010016
      /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
#else
      /* align all transfers to 1 sample */
      err = snd_pcm_sw_params_set_xfer_align (this->audio_fd, this->sw_params, 1);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: Unable to set transfer alignment: %s\n", snd_strerror (err));
        break;
      }
#endif

      /* allow the transfer when at least period_size samples can be processed */
      err = snd_pcm_sw_params_set_avail_min (this->audio_fd, this->sw_params, period_size);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: Unable to set available min: %s\n", snd_strerror (err));
        break;
      }

      /* start the transfer when the buffer contains at least period_size samples */
      err = snd_pcm_sw_params_set_start_threshold (this->audio_fd, this->sw_params, period_size);
      if (err < 0) {
        xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
          "audio_alsa_out: Unable to set start threshold: %s\n", snd_strerror (err));
        break;
      }
    }

    /* never stop the transfer, even on xruns */
    err = snd_pcm_sw_params_set_stop_threshold (this->audio_fd, this->sw_params, this->buffer_size);
    if (err < 0) {
      xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
        "audio_alsa_out: Unable to set stop threshold: %s\n", snd_strerror (err));
      break;
    }

    /* Install swparams into current parameters */
    err = snd_pcm_sw_params (this->audio_fd, this->sw_params);
    if (err < 0) {
      xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
        "audio_alsa_out: Unable to set swparams: %s\n", snd_strerror (err));
      break;
    }

#ifdef ALSA_LOG
    snd_pcm_dump_setup (this->audio_fd, jcd_out);
    snd_pcm_sw_params_dump (this->sw_params, jcd_out);
#endif

    return this->output_sample_rate;
  } while (0);

  snd_pcm_close (this->audio_fd);
  this->audio_fd = NULL;
  return 0;
}

/*
 * Return the number of audio channels
 */
static int ao_alsa_num_channels(ao_driver_t *this_gen) {
  alsa_driver_t *this = (alsa_driver_t *) this_gen;
  return this->num_channels;
}

/*
 * Return the number of bytes per frame
 */
static int ao_alsa_bytes_per_frame(ao_driver_t *this_gen) {
  alsa_driver_t *this = (alsa_driver_t *) this_gen;
  return this->bytes_per_frame;
}

/*
 * Return gap tolerance (in pts)
 */
static int ao_alsa_get_gap_tolerance (ao_driver_t *this_gen) {
  (void)this_gen;
  return GAP_TOLERANCE;
}

/*
 * Return the delay. is frames measured by looking at pending samples
 */
/* FIXME: delay returns invalid data if status is not RUNNING.
 * e.g When there is an XRUN or we are in PREPARED mode.
 */
static int ao_alsa_delay (ao_driver_t *this_gen)  {
  snd_pcm_sframes_t delay = 0;
  int err = 0;
  alsa_driver_t *this = (alsa_driver_t *) this_gen;
#ifdef LOG_DEBUG
  struct timeval now;
  printf("audio_alsa_out:delay:ENTERED\n");
#endif
  err = snd_pcm_delay( this->audio_fd, &delay );

#ifdef LOG_DEBUG
  printf("audio_alsa_out:delay:delay all=%ld err=%d\n",delay, err);
  gettimeofday(&now, 0);
  printf("audio_alsa_out:delay: Time = %ld.%ld\n", now.tv_sec, now.tv_usec);
  printf("audio_alsa_out:delay:FINISHED\n");
#endif

  /*
   * try to recover from errors and recalculate delay
   */
  if(err) {
#ifdef LOG_DEBUG
    printf("gap audio_alsa_out:delay: recovery\n");
#endif
    err = snd_pcm_recover( this->audio_fd, err, 1 );
    err = snd_pcm_delay( this->audio_fd, &delay );
  }

  /*
   * if we have a negative delay try to forward within the buffer
   */
  if(!err && (delay < 0)) {
#ifdef LOG_DEBUG
    printf("gap audio_alsa_out:delay: forwarding frames: %d\n", (int)-delay);
#endif
    err = snd_pcm_forward( this->audio_fd, -delay );
    if(err >= 0) {
      err = snd_pcm_delay( this->audio_fd, &delay );
    }
  }

  /*
   * on error or (still) negative delays ensure delay
   * is not negative
   */
  if (err || (delay < 0))
    delay = 0;

  return delay;
}

#if 0
/*
 * Handle over/under-run
 */
static void xrun(alsa_driver_t *this)
{
  int res;

  /*
     if ((res = snd_pcm_status (this->audio_fd, this->pcm_status)) < 0) {
       printf ("audio_alsa_out: status error: %s\n", snd_strerror(res));
       return;
     }
     snd_pcm_status_dump (this->pcm_status, jcd_out);
  */
  if (snd_pcm_state(this->audio_fd) == SND_PCM_STATE_XRUN) {
    /*
      struct timeval now, diff, tstamp;
      gettimeofday(&now, 0);
      snd_pcm_status_get_trigger_tstamp (this->pcm_status, &tstamp);
      timersub(&now, &tstamp, &diff);
      printf ("audio_alsa_out: xrun!!! (at least %.3f ms long)\n", diff.tv_sec * 1000 + diff.tv_usec / 1000.0);
    */
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "audio_alsa_out: XRUN!!!\n");
    if ((res = snd_pcm_prepare(this->audio_fd))<0) {
      xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "audio_alsa_out: xrun: prepare error: %s", snd_strerror(res));
      return;
    }
    return;         /* ok, data should be accepted again */
  }
}
#endif

/*
 * resume from suspend
 */
static int resume(snd_pcm_t *pcm)
{
  int res;
  while ((res = snd_pcm_resume(pcm)) == -EAGAIN)
    sleep(1);
  if (! res)
    return 0;
  return snd_pcm_prepare(pcm);
}

/*
 * Write audio data to output buffer (blocking using snd_pcm_wait)
 */
static int ao_alsa_write(ao_driver_t *this_gen, int16_t *data, uint32_t count) {
  snd_pcm_sframes_t result;
  snd_pcm_state_t    state;
#ifdef LOG_DEBUG
  struct timeval now;
#endif
  int wait_result;
  int res;
  uint8_t *buffer=(uint8_t *)data;
  snd_pcm_uframes_t number_of_frames = (snd_pcm_uframes_t) count;
  alsa_driver_t *this = (alsa_driver_t *) this_gen;

#ifdef LOG_DEBUG
  printf("audio_alsa_out:write:ENTERED\n");
  gettimeofday(&now, 0);
  printf("audio_alsa_out:write: Time = %ld.%ld\n", now.tv_sec, now.tv_usec);
  printf("audio_alsa_out:write:count=%u\n",count);
#endif
  state = snd_pcm_state(this->audio_fd);
  if (state == SND_PCM_STATE_SUSPENDED) {
    res = resume(this->audio_fd);
    if (res < 0)
      return 0;
    state = snd_pcm_state(this->audio_fd);
  } else if (state == SND_PCM_STATE_DISCONNECTED) {
    /* the device is gone. audio_out.c handles it if we return something < 0 */
    return -1;
  }
  if (state == SND_PCM_STATE_XRUN) {
#ifdef LOG_DEBUG
    printf("audio_alsa_out:write:XRUN before\n");
    snd_pcm_status (this->audio_fd, this->pcm_status);
    snd_pcm_status_dump (this->pcm_status, jcd_out);
#endif
    if ((res = snd_pcm_prepare(this->audio_fd))<0) {
      xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
              "audio_alsa_out: xrun: prepare error: %s", snd_strerror(res));
      return 0;
    }
    state = snd_pcm_state(this->audio_fd);
#ifdef LOG_DEBUG
    printf("audio_alsa_out:write:XRUN after\n");
#endif
  }
  if ( (state != SND_PCM_STATE_PREPARED) &&
       (state != SND_PCM_STATE_RUNNING) &&
       (state != SND_PCM_STATE_DRAINING) ) {
    xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
	    "audio_alsa_out:write:BAD STATE, state = %d\n",state);
  }

  while( number_of_frames > 0) {
    if ( state == SND_PCM_STATE_RUNNING ) {
#ifdef LOG_DEBUG
      printf("audio_alsa_out:write:loop:waiting for Godot\n");
#endif
      snd_pcm_status (this->audio_fd, this->pcm_status);
      if (snd_pcm_status_get_avail (this->pcm_status) < number_of_frames) {
        wait_result = snd_pcm_wait(this->audio_fd, 1000);
#ifdef LOG_DEBUG
        printf("audio_alsa_out:write:loop:wait_result=%d\n",wait_result);
#endif
        if (wait_result <= 0) return 0;
      }
    }
    if (this->mmap != 0) {
      result = snd_pcm_mmap_writei(this->audio_fd, buffer, number_of_frames);
    } else {
      result = snd_pcm_writei(this->audio_fd, buffer, number_of_frames);
    }

    if (result < 0) {
#ifdef LOG_DEBUG
      printf("audio_alsa_out:write:result=%ld:%s\n",result, snd_strerror(result));
#endif
      state = snd_pcm_state(this->audio_fd);
      if (state == SND_PCM_STATE_SUSPENDED) {
	res = resume(this->audio_fd);
	if (res < 0)
	  return 0;
	continue;
      }
      if (state == SND_PCM_STATE_DISCONNECTED) {
        /* the device is gone. audio_out.c handles it if we return something < 0 */
        return -1;
      } else if ( (state != SND_PCM_STATE_PREPARED) &&
           (state != SND_PCM_STATE_RUNNING) &&
           (state != SND_PCM_STATE_DRAINING) ) {
        xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out:write:BAD STATE2, state = %d, going to try XRUN\n",state);
        if ((res = snd_pcm_prepare(this->audio_fd))<0) {
          xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		  "audio_alsa_out: xrun: prepare error: %s", snd_strerror(res));
          return 0;
        }
      }
    }
    if (result > 0) {
      number_of_frames -= result;
      buffer += result * this->bytes_per_frame;
    }
  }
#if 0
  if ( (state == SND_PCM_STATE_RUNNING) ) {
#ifdef LOG_DEBUG
    printf("audio_alsa_out:write:loop:waiting for Godot2\n");
#endif
    wait_result = snd_pcm_wait(this->audio_fd, 1000000);
#ifdef LOG_DEBUG
    printf("audio_alsa_out:write:loop:wait_result=%d\n",wait_result);
#endif
    if (wait_result < 0) return 0;
  }
#endif
#ifdef LOG_DEBUG
  gettimeofday(&now, 0);
  printf("audio_alsa_out:write: Time = %ld.%ld\n", now.tv_sec, now.tv_usec);
  printf("audio_alsa_out:write:FINISHED\n");
#endif
  return 1; /* audio samples were processed ok */
}

/*
 * This is called when the decoder no longer uses the audio
 */
static void ao_alsa_close(ao_driver_t *this_gen) {
  alsa_driver_t *this = (alsa_driver_t *) this_gen;

  if(this->audio_fd) {
    snd_pcm_nonblock(this->audio_fd, 0);
    snd_pcm_drain(this->audio_fd);
    snd_pcm_close(this->audio_fd);
  }
  this->audio_fd = NULL;
  this->has_pause_resume = 0; /* This is set at open time */
}

/*
 * Find out what output modes + capatilities are supported
 */
static uint32_t ao_alsa_get_capabilities (ao_driver_t *this_gen) {
  alsa_driver_t *this = (alsa_driver_t *) this_gen;
  return this->capabilities;
}

/*
 * Shut down audio output driver plugin and free all resources allocated
 */
static void ao_alsa_exit(ao_driver_t *this_gen) {
  alsa_driver_t *this = (alsa_driver_t *) this_gen;

  this->class->xine->config->unregister_callbacks (this->class->xine->config,
    NULL, NULL, this, sizeof (*this));

  /*
   * Destroy the mixer thread and cleanup the mixer, so that
   * any child processes (such as xscreensaver) cannot inherit
   * the mixer's handle and keep it open.
   * By rejoining the mixer thread, we remove a race condition
   * between closing the handle and spawning the child process
   * (i.e. xscreensaver).
   */

  if(this->mixer.handle && this->mixer.thread_created) {
    this->mixer.running = 0;
    pthread_join(this->mixer.thread, NULL);
    snd_mixer_close(this->mixer.handle);
    this->mixer.handle=0;
  }
  pthread_mutex_destroy(&this->mixer.mutex);

  if (this->audio_fd) snd_pcm_close(this->audio_fd);
  this->audio_fd=NULL;

  xine_config_free_string(this->class->xine, &this->mixer.name);

  {
    uint32_t u;
    for (u = 0; u < sizeof (this->devs) / sizeof (this->devs[0]); u++) {
      _x_freep (&this->devs[u].name);
    }
  }

  free (this);
}

/*
 * Get a property of audio driver
 */
static int ao_alsa_get_property (ao_driver_t *this_gen, int property) {
  alsa_driver_t *this = (alsa_driver_t *) this_gen;
  int err;

  switch(property) {
  case AO_PROP_MIXER_VOL:
  case AO_PROP_PCM_VOL:
    if(this->mixer.elem) {
      int vol;

      pthread_mutex_lock(&this->mixer.mutex);

      if((err = snd_mixer_selem_get_playback_volume(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT,
						    &this->mixer.left_vol)) < 0) {
	xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: snd_mixer_selem_get_playback_volume(): %s\n",  snd_strerror(err));
	goto done;
      }

      if((err = snd_mixer_selem_get_playback_volume(this->mixer.elem, SND_MIXER_SCHN_FRONT_RIGHT,
						    &this->mixer.right_vol)) < 0) {
	xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: snd_mixer_selem_get_playback_volume(): %s\n",  snd_strerror(err));
	goto done;
      }

    done:
      vol = (((ao_alsa_get_percent_from_volume(this->mixer.left_vol, this->mixer.min, this->mixer.max)) +
	      (ao_alsa_get_percent_from_volume(this->mixer.right_vol, this->mixer.min, this->mixer.max))) /2);
      pthread_mutex_unlock(&this->mixer.mutex);

      return vol;
    }
    break;

  case AO_PROP_MUTE_VOL:
    {
      int mute;

      pthread_mutex_lock(&this->mixer.mutex);
      mute = ((this->mixer.mute & MIXER_HAS_MUTE_SWITCH) && (this->mixer.mute & MIXER_MASK_MUTE)) ? 1 : 0;
      pthread_mutex_unlock(&this->mixer.mutex);

      return mute;
    }
    break;
  }

  return 0;
}

/*
 * Set a property of audio driver
 */
static int ao_alsa_set_property (ao_driver_t *this_gen, int property, int value) {
  alsa_driver_t *this = (alsa_driver_t *) this_gen;
  int err;

  switch(property) {
  case AO_PROP_MIXER_VOL:
  case AO_PROP_PCM_VOL:
    if(this->mixer.elem) {

      pthread_mutex_lock(&this->mixer.mutex);

      this->mixer.left_vol = this->mixer.right_vol = ao_alsa_get_volume_from_percent(value, this->mixer.min, this->mixer.max);

      if((err = snd_mixer_selem_set_playback_volume(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT,
						    this->mixer.left_vol)) < 0) {
	xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: snd_mixer_selem_get_playback_volume(): %s\n",  snd_strerror(err));
	pthread_mutex_unlock(&this->mixer.mutex);
	return ~value;
      }

      if((err = snd_mixer_selem_set_playback_volume(this->mixer.elem, SND_MIXER_SCHN_FRONT_RIGHT,
						    this->mixer.right_vol)) < 0) {
	xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: snd_mixer_selem_get_playback_volume(): %s\n",  snd_strerror(err));
	pthread_mutex_unlock(&this->mixer.mutex);
	return ~value;
      }
      pthread_mutex_unlock(&this->mixer.mutex);
      return value;
    }
    break;

  case AO_PROP_MUTE_VOL:
    if(this->mixer.elem) {

      if(this->mixer.mute & MIXER_HAS_MUTE_SWITCH) {
	int swl = 0, swr = 0;
	int old_mute;

	pthread_mutex_lock(&this->mixer.mutex);

	old_mute = this->mixer.mute;
	if(value)
	  this->mixer.mute |= MIXER_MASK_MUTE;
	else
	  this->mixer.mute &= ~MIXER_MASK_MUTE;

	if ((this->mixer.mute & MIXER_MASK_MUTE) != (old_mute & MIXER_MASK_MUTE)) {
	  if(this->mixer.mute & MIXER_MASK_STEREO) {
	    snd_mixer_selem_get_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT, &swl);
	    snd_mixer_selem_set_playback_switch_all(this->mixer.elem, !swl);
	  }
	  else {
	    if (this->mixer.mute & MIXER_MASK_LEFT) {
	      snd_mixer_selem_get_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT, &swl);
	      snd_mixer_selem_set_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT, !swl);
	    }
	    if (SND_MIXER_SCHN_FRONT_RIGHT != SND_MIXER_SCHN_UNKNOWN && (this->mixer.mute & MIXER_MASK_RIGHT)) {
	      snd_mixer_selem_get_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_RIGHT, &swr);
	      snd_mixer_selem_set_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_RIGHT, !swr);
	    }
	  }
	}

	pthread_mutex_unlock(&this->mixer.mutex);
      }
      return value;
    }
    return ~value;
    break;
  }

  return ~value;
}

/*
 * Misc control operations
 */
static int ao_alsa_ctrl(ao_driver_t *this_gen, int cmd, ...) {
  alsa_driver_t *this = (alsa_driver_t *) this_gen;
  int err;

  /* Alsa 0.9.x pause and resume is not stable enough at the moment.
   * Use snd_pcm_drop and restart instead.
   */
  switch (cmd) {

  case AO_CTRL_PLAY_PAUSE:
    if (this->audio_fd) {
      if (this->has_pause_resume) {
        if ((err=snd_pcm_pause(this->audio_fd, 1)) < 0) {
          xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		  "audio_alsa_out: Pause call failed. (err=%d:%s)\n",err, snd_strerror(err));
          this->has_pause_resume = 0;
          ao_alsa_ctrl(this_gen, AO_CTRL_PLAY_PAUSE, NULL);
        } else {
          this->is_paused = 1;
	}
      } else {
        if ((err=snd_pcm_reset(this->audio_fd)) < 0) {
          xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		  "audio_alsa_out: Reset call failed. (err=%d:%s)\n",err, snd_strerror(err));
        }
        if ((err=snd_pcm_drain(this->audio_fd)) < 0) {
          xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		  "audio_alsa_out: Drain call failed. (err=%d:%s)\n",err, snd_strerror(err));
        }
        if ((err=snd_pcm_prepare(this->audio_fd)) < 0) {
          xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		  "audio_alsa_out: Prepare call failed. (err=%d:%s)\n",err, snd_strerror(err));
        }
      }
    }
    break;

  case AO_CTRL_PLAY_RESUME:
    if (this->audio_fd) {
      if (this->has_pause_resume && this->is_paused) {
        if ((err=snd_pcm_pause(this->audio_fd, 0)) < 0) {
          if (err == -77) {
            xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		    "audio_alsa_out: Warning: How am I supposed to RESUME, if I am not PAUSED. "
		    "audio_out.c, please don't call me!\n");
            break;
          }
          xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		  "audio_alsa_out: Resume call failed. (err=%d:%s)\n",err, snd_strerror(err));
          this->has_pause_resume = 0;
        } else {
          this->is_paused = 0;
	}
      }
    }
    break;

  case AO_CTRL_FLUSH_BUFFERS:
    if (this->audio_fd) {
      if ((err=snd_pcm_drop(this->audio_fd)) < 0) {
        xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: Drop call failed. (err=%d:%s)\n",err, snd_strerror(err));
      }
      if ((err=snd_pcm_prepare(this->audio_fd)) < 0) {
        xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: Prepare call failed. (err=%d:%s)\n",err, snd_strerror(err));
      }
    }
    break;
  }

  return 0;
}

/*
 * Initialize mixer
 */
static void ao_alsa_mixer_init(ao_driver_t *this_gen) {
  alsa_driver_t        *this = (alsa_driver_t *) this_gen;
  config_values_t      *config = this->class->xine->config;
  const char           *pcm_device;
  snd_ctl_t            *ctl_handle;
  int                   err;
  void                 *mixer_sid;
  snd_mixer_elem_t     *elem;
  int                   mixer_n_selems = 0;
  snd_mixer_selem_id_t *sid;
  int                   loop = 0;
  int                   found;
  int                   swl = 0, swr = 0, send_events;

  this->mixer.elem = 0;
  pcm_device = this->devs[0].name;
  err = snd_ctl_open (&ctl_handle, pcm_device, 0);
  if (err < 0) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "audio_alsa_out: snd_ctl_open(): %s\n", snd_strerror(err));
    return;
  }

  if ((err = snd_ctl_card_info (ctl_handle, this->card_info)) < 0) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
	     "audio_alsa_out: snd_ctl_card_info(): %s\n", snd_strerror(err));
    snd_ctl_close(ctl_handle);
    return;
  }

  snd_ctl_close (ctl_handle);

  /*
   * Open mixer device
   */
  if ((err = snd_mixer_open (&this->mixer.handle, 0)) < 0) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
	     "audio_alsa_out: snd_mixer_open(): %s\n", snd_strerror(err));
    this->mixer.handle=0;
    return;
  }

  if ((err = snd_mixer_attach (this->mixer.handle, pcm_device)) < 0) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
	     "audio_alsa_out: snd_mixer_attach(): %s\n", snd_strerror(err));
    snd_mixer_close(this->mixer.handle);
    this->mixer.handle=0;
    return;
  }

  if ((err = snd_mixer_selem_register (this->mixer.handle, NULL, NULL)) < 0) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
	     "audio_alsa_out: snd_mixer_selem_register(): %s\n", snd_strerror(err));
    snd_mixer_close(this->mixer.handle);
    this->mixer.handle=0;
    return;
  }

  if ((err = snd_mixer_load (this->mixer.handle)) < 0) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
	     "audio_alsa_out: snd_mixer_load(): %s\n", snd_strerror(err));
    snd_mixer_close(this->mixer.handle);
    this->mixer.handle=0;
    return;
  }

  mixer_sid = calloc (1, snd_mixer_selem_id_sizeof () * snd_mixer_get_count (this->mixer.handle));
  if (mixer_sid == NULL) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
      "audio_alsa_out: malloc () failed: %s\n", strerror (errno));
    snd_mixer_close(this->mixer.handle);
    this->mixer.handle=0;
    return;
  }

 again:

  found = 0;
  mixer_n_selems = 0;
  for (elem = snd_mixer_first_elem(this->mixer.handle); elem; elem = snd_mixer_elem_next(elem)) {
    sid = (snd_mixer_selem_id_t *)(((char *)mixer_sid) + snd_mixer_selem_id_sizeof() * mixer_n_selems);

    if ((snd_mixer_elem_get_type(elem) != SND_MIXER_ELEM_SIMPLE) ||
        !snd_mixer_selem_is_active(elem))
      continue;

    snd_mixer_selem_get_id(elem, sid);
    mixer_n_selems++;

    if(!strcmp((snd_mixer_selem_get_name(elem)), this->mixer.name)) {
      /* printf("found %s\n", snd_mixer_selem_get_name(elem)); */

      this->mixer.elem = elem;

      snd_mixer_selem_get_playback_volume_range(this->mixer.elem,
						&this->mixer.min, &this->mixer.max);
      if((err = snd_mixer_selem_get_playback_volume(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT,
						    &this->mixer.left_vol)) < 0) {
	xprintf(this->class->xine, XINE_VERBOSITY_DEBUG,
		"audio_alsa_out: snd_mixer_selem_get_playback_volume(): %s\n",  snd_strerror(err));
	this->mixer.elem = NULL;
	continue;
      }

      if((err = snd_mixer_selem_get_playback_volume(this->mixer.elem, SND_MIXER_SCHN_FRONT_RIGHT,
						    &this->mixer.right_vol)) < 0) {
	xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
		 "audio_alsa_out: snd_mixer_selem_get_playback_volume(): %s\n",  snd_strerror(err));
	this->mixer.elem = NULL;
	continue;
      }

      /* Channels mute */
      this->mixer.mute = 0;
      if(snd_mixer_selem_has_playback_switch(this->mixer.elem)) {
	this->mixer.mute |= MIXER_HAS_MUTE_SWITCH;

	if (snd_mixer_selem_has_playback_switch_joined(this->mixer.elem)) {
	  this->mixer.mute |= MIXER_MASK_STEREO;
	  snd_mixer_selem_get_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT, &swl);
	}
	else {
	  this->mixer.mute |= MIXER_MASK_LEFT;
	  snd_mixer_selem_get_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_LEFT, &swl);

	  if (SND_MIXER_SCHN_FRONT_RIGHT != SND_MIXER_SCHN_UNKNOWN) {
	    this->mixer.mute |= MIXER_MASK_RIGHT;
	    snd_mixer_selem_get_playback_switch(this->mixer.elem, SND_MIXER_SCHN_FRONT_RIGHT, &swr);
	  }

	  if(!swl || !swr)
	    this->mixer.mute |= MIXER_MASK_MUTE;
	}

	this->capabilities |= AO_CAP_MUTE_VOL;
      }

      found++;

      goto mixer_found;
    }
  }

  if(loop)
    goto mixer_found; /* Yes, untrue but... ;-) */

  if(!strcmp(this->mixer.name, "PCM")) {
    config->update_string(config, "audio.device.alsa_mixer_name", "Master");
    loop++;
  }
  else {
    config->update_string(config, "audio.device.alsa_mixer_name", "PCM");
  }

  config->free_string(config, &this->mixer.name);
  this->mixer.name = config->lookup_string(config, "audio.device.alsa_mixer_name");

  goto again;

 mixer_found:

  free (mixer_sid);

  /*
   * Ugly: yes[*]  no[ ]
   */
  if(found) {
    if(!strcmp(this->mixer.name, "Master"))
      this->capabilities |= AO_CAP_MIXER_VOL;
    else
      this->capabilities |= AO_CAP_PCM_VOL;
  } else {
    if (this->mixer.handle) {
      snd_mixer_close(this->mixer.handle);
      this->mixer.handle=0;
    }
    return;
  }

  /* Create a thread which wait/handle mixer events */
  send_events = config->register_bool(config, "audio.alsa_hw_mixer", 1,
				      _("notify changes to the hardware mixer"),
				      _("When the hardware mixer changes, your application will receive "
				        "a notification so that it can update its graphical representation "
				        "of the mixer settings on the fly."),
				      10, NULL, NULL);

  if (send_events && found) {
    pthread_attr_t       pth_attrs;
    struct sched_param   pth_params;

    this->mixer.running = 1;

    pthread_attr_init(&pth_attrs);

    pthread_attr_getschedparam(&pth_attrs, &pth_params);
    pth_params.sched_priority = sched_get_priority_min(SCHED_OTHER);
    pthread_attr_setschedparam(&pth_attrs, &pth_params);
    if (pthread_create(&this->mixer.thread, &pth_attrs, ao_alsa_handle_event_thread, (void *) this)) {
      xprintf (this->class->xine, XINE_VERBOSITY_LOG, LOG_MODULE ": pthread_create() failed\n");
    } else {
      this->mixer.thread_created = 1;
    }
    pthread_attr_destroy(&pth_attrs);
  }
}

static void alsa_apply_speaker_arrangement (alsa_driver_t *this, int speakers) {
  char logbuf[2048], *q, *logend = logbuf + sizeof (logbuf);

  q = logbuf;
  q += strlcpy (q, _("audio_alsa_out : supported modes are"), logend - q);
  if (q >= logend)
    q = logend;

  if (this->capabilities & AO_CAP_8BITS) {
    q += strlcpy (q, this->bits_names[0], logend - q);
    if (q >= logend)
      q = logend;
  }
  if (this->capabilities & AO_CAP_16BITS) {
    q += strlcpy (q, this->bits_names[1], logend - q);
    if (q >= logend)
      q = logend;
  }
  if (this->capabilities & AO_CAP_24BITS) {
    q += strlcpy (q, this->bits_names[2], logend - q);
    if (q >= logend)
      q = logend;
  }
  if (this->capabilities & AO_CAP_FLOAT32) {
    q += strlcpy (q, this->bits_names[3], logend - q);
    if (q >= logend)
      q = logend;
  }

  this->capabilities &= ~(AO_CAP_MODE_MONO |
                          AO_CAP_MODE_STEREO |
                          AO_CAP_MODE_4CHANNEL |
                          AO_CAP_MODE_4_1CHANNEL |
                          AO_CAP_MODE_5CHANNEL |
                          AO_CAP_MODE_5_1CHANNEL |
                          AO_CAP_MODE_A52 |
                          AO_CAP_MODE_AC5);

  /* always set these. */
  if (this->supported_channels & (1 << 1)) {
    this->capabilities |= AO_CAP_MODE_MONO;
    q += strlcpy (q, this->devs[0].type, logend - q);
    if (q >= logend)
      q = logend;
  }
  if (this->supported_channels & (1 << 2)) {
    this->capabilities |= AO_CAP_MODE_STEREO;
    q += strlcpy (q, this->devs[1].type, logend - q);
    if (q >= logend)
      q = logend;
  }

  if (this->supported_channels & (1 << 4)) {
    const char *t;
    if (speakers == SURROUND4) {
      this->capabilities |= AO_CAP_MODE_4CHANNEL;
      t = this->devs[2].type;
    } else {
      t = _(" (4-channel not enabled in xine config)");
    }
    q += strlcpy (q, t, logend - q);
    if (q >= logend)
      q = logend;
  }
  if (this->supported_channels & (1 << 6)) {
    const char *t;
    if (speakers == SURROUND41) {
      this->capabilities |= AO_CAP_MODE_4_1CHANNEL;
      t = _(" 4.1-channel");
    } else {
      t = _(" (4.1-channel not enabled in xine config)");
    }
    q += strlcpy (q, t, logend - q);
    if (q >= logend)
      q = logend;
    if (speakers == SURROUND5) {
      this->capabilities |= AO_CAP_MODE_5CHANNEL;
      t = _(" 5-channel");
    } else {
      t = _(" (5-channel not enabled in xine config)");
    }
    q += strlcpy (q, t, logend - q);
    if (q >= logend)
      q = logend;
    /* NOTE: That ">=" covers both the high channel counts and passthrough mode.
     * This is more or less a BUG when a traditional SPDIF link is used.
     * However, if alsa_surround51_device refers to the same HDMI port
     * as alsa_passthrough_device, then we have a HACK that routes software
     * decoded 5.1 where it belongs :-) */
    if (speakers >= SURROUND51) {
      this->capabilities |= AO_CAP_MODE_5_1CHANNEL;
      t = this->devs[3].type;
    } else {
      t = _(" (5.1-channel not enabled in xine config)");
    }
    q += strlcpy (q, t, logend - q);
    if (q >= logend)
      q = logend;
  }

  {
    const char *t;
    if (speakers == A52_PASSTHRU) {
      this->capabilities |= AO_CAP_MODE_A52 | AO_CAP_MODE_AC5;
      t = this->devs[4].type;
    } else {
      t = _(" (a/52 and DTS pass-through not enabled in xine config)");
    }
    q += strlcpy (q, t, logend - q);
/*  if (q >= logend)
      q = logend; */
  }

  xprintf (this->class->xine, XINE_VERBOSITY_LOG, "%s.\n", logbuf);
}

static void _alsa_mmap_cb (void *user_data, xine_cfg_entry_t *entry) {
  alsa_driver_t *this = (alsa_driver_t *)user_data;
  this->mmap = entry->num_value;
}

static void _alsa_speaker_arrangement_cb (void *user_data, xine_cfg_entry_t *entry) {
  alsa_driver_t *this = (alsa_driver_t *)user_data;
  alsa_apply_speaker_arrangement (this, entry->num_value);
}

static char *_alsa_safe_strdup (const char *s) {
  return s ? strdup (s) : NULL;
}

static void _alsa_dev_name_cb (void *user_data, xine_cfg_entry_t *entry) {
  struct _alsa_dev_info_s *info = (struct _alsa_dev_info_s *)user_data;
  free (info->name);
  info->name = _alsa_safe_strdup (entry->str_value);
}

static int _alsa_query_dev (alsa_driver_t *this, uint32_t index) {
  char logbuf[2048], *q = logbuf, *logend = logbuf + sizeof (logbuf);
  struct _alsa_dev_info_s *info = &this->devs[index];
  const char *name;
  uint32_t u;
  int err;

  name = info->name ? info->name : "default";
  if (1) {
    /* skip doubles */
    for (u = 0; u < sizeof (this->devs) / sizeof (this->devs[0]); u++) {
      if (u != index) {
        const char *alias = this->devs[u].name ? this->devs[u].name : "default";
        if (!strcmp (name, alias) && this->devs[u].supported_channels) {
          info->capabilities = this->devs[u].capabilities;
          info->supported_channels = this->devs[u].supported_channels;
          xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
            "audio_alsa_out: already probed \"%s\" for%s.\n", name, info->type);
          return 2;
        }
      }
    }
  }
  xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
    "audio_alsa_out: probing \"%s\" for%s ...\n", name, info->type);
  q += strlcpy (q, "audio_alsa_out: ", logend - q);
  if (q > logend)
    q = logend;
  q += strlcpy (q, name, logend - q);
  if (q > logend)
    q = logend;
  q += strlcpy (q, ": ", logend - q);
  if (q > logend)
    q = logend;

  err = snd_pcm_open (&this->audio_fd, name, SND_PCM_STREAM_PLAYBACK, 1); /* NON-BLOCK mode */
  if (err < 0) {
    xine_log (this->class->xine, XINE_LOG_MSG,
      _("snd_pcm_open() failed:%d:%s\n"), err, snd_strerror (err));
    xine_log (this->class->xine, XINE_LOG_MSG,
      _(">>> Check if another program already uses PCM <<<\n"));
    return 0;
  }

  err = snd_pcm_hw_params_any (this->audio_fd, this->hw_params);
  if (err < 0) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
      "audio_alsa_out: broken configuration for this PCM: no configurations available.\n");
    snd_pcm_close (this->audio_fd);
    this->audio_fd = NULL;
    return 0;
  }

  info->capabilities &= ~(AO_CAP_8BITS | AO_CAP_16BITS | AO_CAP_24BITS | AO_CAP_FLOAT32);
  if (!(snd_pcm_hw_params_test_format (this->audio_fd, this->hw_params, SND_PCM_FORMAT_U8))) {
    info->capabilities |= AO_CAP_8BITS;
    q += strlcpy (q, this->bits_names[0], logend - q);
    if (q > logend)
      q = logend;
  }
  if (!(snd_pcm_hw_params_test_format (this->audio_fd, this->hw_params, SND_PCM_FORMAT_S16))) {
    info->capabilities |= AO_CAP_16BITS;
    q += strlcpy (q, this->bits_names[1], logend - q);
    if (q > logend)
      q = logend;
  }
  if (!(snd_pcm_hw_params_test_format (this->audio_fd, this->hw_params, SND_PCM_FORMAT_S24))) {
    info->capabilities |= AO_CAP_24BITS;
    q += strlcpy (q, this->bits_names[2], logend - q);
    if (q > logend)
      q = logend;
  }
  if (!(snd_pcm_hw_params_test_format (this->audio_fd, this->hw_params, SND_PCM_FORMAT_FLOAT))) {
    info->capabilities |= AO_CAP_FLOAT32;
    q += strlcpy (q, this->bits_names[3], logend - q);
    if (q > logend)
      q = logend;
  }
  if (0 == (info->capabilities & (AO_CAP_FLOAT32 | AO_CAP_24BITS | AO_CAP_16BITS | AO_CAP_8BITS))) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "audio_alsa_out: no supported PCM format found.\n");
    snd_pcm_close (this->audio_fd);
    this->audio_fd = NULL;
    return 0;
  }

  info->supported_channels = 0;
  if (!(snd_pcm_hw_params_test_channels (this->audio_fd, this->hw_params, 1))) {
    info->supported_channels |= 1 << 1;
    q += strlcpy (q, this->devs[0].type, logend - q);
    if (q > logend)
      q = logend;
  }
  if (!(snd_pcm_hw_params_test_channels (this->audio_fd, this->hw_params, 2))) {
    info->supported_channels |= 1 << 2;
    q += strlcpy (q, this->devs[1].type, logend - q);
    if (q > logend)
      q = logend;
  }
  if (!(snd_pcm_hw_params_test_channels (this->audio_fd, this->hw_params, 4))) {
    info->supported_channels |= 1 << 4;
    q += strlcpy (q, this->devs[2].type, logend - q);
    if (q > logend)
      q = logend;
  }
  if (!(snd_pcm_hw_params_test_channels (this->audio_fd, this->hw_params, 6))) {
    info->supported_channels |= 1 << 6;
    q += strlcpy (q, this->devs[3].type, logend - q);
    if (q > logend)
      q = logend;
  }

  err = snd_pcm_hw_params_set_access (this->audio_fd, this->hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
  if (err < 0) {
    xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
      "audio_alsa_out: interleaved access not available.");
    snd_pcm_close (this->audio_fd);
    this->audio_fd = NULL;
    return 0;
  }

  if (index == 4) {
    info->capabilities |= AO_CAP_MODE_A52 | AO_CAP_MODE_AC5;
    q += strlcpy (q, this->devs[4].type, logend - q);
    if (q > logend)
      q = logend;
  }

  xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "%s.\n", logbuf);

  this->supported_channels = 0;
  for (u = 0; u < sizeof (this->devs) / sizeof (this->devs[0]); u++) {
    this->supported_channels |= this->devs[u].supported_channels;
    if (u != index) {
      const char *alias = this->devs[u].name ? this->devs[u].name : "default";
      if (!strcmp (name, alias)) {
        this->devs[u].capabilities = info->capabilities;
        this->devs[u].supported_channels = info->supported_channels;
      }
    }
  }

  snd_pcm_close (this->audio_fd);
  this->audio_fd = NULL;
  return 1;
}

/*
 * Initialize plugin
 */

static ao_driver_t *open_plugin (audio_driver_class_t *class_gen, const void *data) {
  alsa_class_t        *class = (alsa_class_t *) class_gen;
  config_values_t     *config = class->xine->config;
  alsa_driver_t       *this;

  (void)data;
  {
    uint32_t s_this      = (sizeof (*this) + 15) & ~15u;
    uint32_t s_hw_params = (snd_pcm_hw_params_sizeof () + 15) & ~15u;
    uint32_t s_sw_params = (snd_pcm_sw_params_sizeof () + 15) & ~15u;
    uint32_t s_ac_mask   = (snd_pcm_access_mask_sizeof () + 15) & ~15u;
    uint32_t s_card_info = (snd_ctl_card_info_sizeof () + 15) & ~15u;
    uint32_t s_status    = (snd_pcm_status_sizeof () + 15) & ~15u;
    uint8_t *m = calloc (1, s_this + s_hw_params + s_sw_params + s_ac_mask + s_card_info + s_status);
    if (!m)
      return NULL;
    this = (alsa_driver_t *)m;
    m += s_this;
    this->hw_params  = (snd_pcm_hw_params_t *)m;
    m += s_hw_params;
    this->sw_params  = (snd_pcm_sw_params_t *)m;
    m += s_sw_params;
    this->ac_mask    = (snd_pcm_access_mask_t *)m;
    m += s_ac_mask;
    this->card_info  = (snd_ctl_card_info_t *)m;
    m += s_card_info;
    this->pcm_status = (snd_pcm_status_t *)m;
  }
#ifndef HAVE_ZERO_SAFE_MEM
  this->devs[0].supported_channels = 0;
  this->devs[0].capabilities       = 0;
  this->devs[1].supported_channels = 0;
  this->devs[1].capabilities       = 0;
  this->devs[2].supported_channels = 0;
  this->devs[2].capabilities       = 0;
  this->devs[3].supported_channels = 0;
  this->devs[3].capabilities       = 0;
  this->devs[4].supported_channels = 0;
  this->devs[4].capabilities       = 0;
  this->capabilities               = 0;
  this->has_pause_resume           = 0; /* This is checked at open time instead */
  this->is_paused                  = 0;
  this->output_sample_rate         = 0;
#endif

  this->class = class;

  this->bits_names[0] = _(" 8bit");
  this->bits_names[1] = _(" 16bit");
  this->bits_names[2] = _(" 24bit");
  this->bits_names[3] = _(" 32bit");

  this->mmap = config->register_bool (config, "audio.device.alsa_mmap_enable", 0,
    _("sound card can do mmap"),
    _("Enable this, if your sound card and alsa driver support memory mapped IO.\n"
      "You can try enabling it and check, if everything works. If it does, this "
      "will increase performance."),
    10, _alsa_mmap_cb, this);

  this->devs[0].this = this;
  this->devs[0].type = _(" mono");
  this->devs[0].config_key = "audio.device.alsa_default_device";
  this->devs[0].name = _alsa_safe_strdup (config->register_string (config,
    this->devs[0].config_key,
    "default",
    _("device used for mono output"),
    _("xine will use this alsa device to output mono sound.\n"
      "See the alsa documentation for information on alsa devices."),
    10, _alsa_dev_name_cb, &this->devs[0]));

  this->devs[1].this = this;
  this->devs[1].type = _(" stereo");
  this->devs[1].config_key = "audio.device.alsa_front_device";
  this->devs[1].name = _alsa_safe_strdup (config->register_string (config,
    this->devs[1].config_key,
    "plug:front:default",
    _("device used for stereo output"),
    _("xine will use this alsa device to output stereo sound.\n"
      "See the alsa documentation for information on alsa devices."),
    10, _alsa_dev_name_cb, &this->devs[1]));

  this->devs[2].this = this;
  this->devs[2].type = _(" 4-channel");
  this->devs[2].config_key = "audio.device.alsa_surround40_device";
  this->devs[2].name = _alsa_safe_strdup (config->register_string (config,
    this->devs[2].config_key,
    "plug:surround40:0",
    _("device used for 4-channel output"),
    _("xine will use this alsa device to output 4 channel (4.0) surround sound.\n"
      "See the alsa documentation for information on alsa devices."),
    10, _alsa_dev_name_cb, &this->devs[2]));

  this->devs[3].this = this;
  this->devs[3].type = _(" 5.1-channel");
  this->devs[3].config_key = "audio.device.alsa_surround51_device";
  this->devs[3].name = _alsa_safe_strdup (config->register_string (config,
    this->devs[3].config_key,
    "plug:surround51:0",
    _("device used for 5.1-channel output"),
    _("xine will use this alsa device to output 5 channel plus LFE (5.1) surround sound.\n"
      "See the alsa documentation for information on alsa devices."),
    10, _alsa_dev_name_cb, &this->devs[3]));

  this->devs[4].this = this;
  this->devs[4].type = _(" a/52 and DTS pass-through");
  this->devs[4].config_key = "audio.device.alsa_passthrough_device";
  this->devs[4].name = _alsa_safe_strdup (config->register_string (config,
    this->devs[4].config_key,
    "iec958:AES0=0x6,AES1=0x82,AES2=0x0,AES3=0x2",
    _("device used for a/52 and DTS pass-through"),
    _("xine will use this alsa device to output undecoded digital surround sound. "
      "This can be used be external surround decoders.\n"
      "See the alsa documentation for information on alsa devices."),
    10, _alsa_dev_name_cb, &this->devs[4]));

  {
    int err = snd_lib_error_set_handler (error_callback);
    if (err < 0)
      xine_log (this->class->xine, XINE_LOG_MSG, _("snd_lib_error_set_handler() failed: %d"), err);
  }

  {
    AUDIO_DEVICE_SPEAKER_ARRANGEMENT_TYPES;
    int speakers;

    /* for usability reasons, keep this in sync with audio_oss_out.c */
    speakers = config->register_enum (config,
      "audio.output.speaker_arrangement",
      STEREO,
      (char **)speaker_arrangement,
      AUDIO_DEVICE_SPEAKER_ARRANGEMENT_HELP,
      0, _alsa_speaker_arrangement_cb, this);

    /* query the devices we might need, and stop at stereo. */
    switch (speakers) {
      case A52_PASSTHRU:
        if (_alsa_query_dev (this, 4))
          this->capabilities |= this->devs[4].capabilities;
        /* fall through */
      case SURROUND71:
      case SURROUND61:
      case SURROUND6:
      case SURROUND51:
      case SURROUND5:
      case SURROUND41:
        if (_alsa_query_dev (this, 3)) {
          this->capabilities |= this->devs[3].capabilities;
          break;
        }
        /* fall through */
      case SURROUND4:
        if (_alsa_query_dev (this, 2))
          this->capabilities |= this->devs[2].capabilities;
        break;
      default: ;
    }
    if (_alsa_query_dev (this, 0))
      this->capabilities |= this->devs[0].capabilities;
    if (_alsa_query_dev (this, 1)) {
      this->capabilities |= this->devs[1].capabilities;
    } else {
      /* Fallback to "default" if device "front" does not exist.
       * Needed for some very basic sound cards. */
      config->update_string (config, this->devs[1].config_key, "default");
      if (_alsa_query_dev (this, 1))
        this->capabilities |= this->devs[1].capabilities;
    }

    alsa_apply_speaker_arrangement (this, speakers);
  }

  do {
    /* printf("audio_alsa_out: capabilities 0x%X\n",this->capabilities); */
    if (!this->capabilities)
      break;

    config->register_string (config,
      "audio.device.alsa_mixer_name",
      "PCM",
      _("alsa mixer device"),
      _("xine will use this alsa mixer device to change the volume.\n"
        "See the alsa documentation for information on alsa devices."),
      10, NULL, NULL);
    this->mixer.name = config->lookup_string(config, "audio.device.alsa_mixer_name");
    if (!this->mixer.name)
      break;

    pthread_mutex_init (&this->mixer.mutex, NULL);
    ao_alsa_mixer_init (&this->ao_driver);

    this->ao_driver.get_capabilities    = ao_alsa_get_capabilities;
    this->ao_driver.get_property        = ao_alsa_get_property;
    this->ao_driver.set_property        = ao_alsa_set_property;
    this->ao_driver.open                = ao_alsa_open;
    this->ao_driver.num_channels        = ao_alsa_num_channels;
    this->ao_driver.bytes_per_frame     = ao_alsa_bytes_per_frame;
    this->ao_driver.delay               = ao_alsa_delay;
    this->ao_driver.write               = ao_alsa_write;
    this->ao_driver.close               = ao_alsa_close;
    this->ao_driver.exit                = ao_alsa_exit;
    this->ao_driver.get_gap_tolerance   = ao_alsa_get_gap_tolerance;
    this->ao_driver.control             = ao_alsa_ctrl;

    return &this->ao_driver;
  } while (0);

  config->unregister_callbacks (config, NULL, NULL, this, sizeof (*this));
  {
    uint32_t u;
    for (u = 0; u < sizeof (this->devs) / sizeof (this->devs[0]); u++) {
      _x_freep (&this->devs[u].name);
    }
  }
  free (this);
  return NULL;
}


/*
 * class functions
 */
static void *init_class (xine_t *xine, const void *data) {
  alsa_class_t *this;

  (void)data;
  this = calloc(1, sizeof (alsa_class_t));
  if (!this)
    return NULL;

  this->driver_class.open_plugin     = open_plugin;
  this->driver_class.identifier      = "alsa";
  this->driver_class.description     = N_("xine audio output plugin using alsa-compliant audio devices/drivers");
  this->driver_class.dispose         = default_audio_driver_class_dispose;

/*  this->config = xine->config; */
  this->xine = xine;
  return this;
 }

static ao_info_t ao_info_alsa = {
  .priority = 10,
};

/*
 * exported plugin catalog entry
 */

const plugin_info_t xine_plugin_info[] EXPORTED = {
  /* type, API, "name", version, special_info, init_function */
  { PLUGIN_AUDIO_OUT, AO_OUT_ALSA_IFACE_VERSION, "alsa", XINE_VERSION_CODE, &ao_info_alsa, init_class },
  { PLUGIN_NONE, 0, NULL, 0, NULL, NULL }
};