Rename directories, move files (step 1 of 2) (does not compile)

Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.

Renames the following directories:
    libaf -> audio/filter
    libao2 -> audio/out
    libvo -> video/out
    libmpdemux -> demux

Split libmpcodecs:
    vf* -> video/filter
    vd*, dec_video.* -> video/decode
    mp_image*, img_format*, ... -> video/
    ad*, dec_audio.* -> audio/decode

libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.

Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.

sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).

Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.

1 files changed, 0 insertions, 868 deletions

diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c
deleted file mode 100644
index 27119112cb..0000000000
--- a/libao2/ao_alsa.c
+++ /dev/null
@@ -1,868 +0,0 @@
-/*
- * ALSA 0.9.x-1.x audio output driver
- *
- * Copyright (C) 2004 Alex Beregszaszi
- *
- * modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de>
- * additional AC-3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org>
- * 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.
- *
- * This file is part of MPlayer.
- *
- * MPlayer 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.
- *
- * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <errno.h>
-#include <sys/time.h>
-#include <stdlib.h>
-#include <stdarg.h>
-#include <ctype.h>
-#include <math.h>
-#include <string.h>
-#include <alloca.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
-
-#include <alsa/asoundlib.h>
-
-#include "audio_out.h"
-#include "audio_out_internal.h"
-#include "libaf/format.h"
-
-static const ao_info_t info =
-{
-    "ALSA-0.9.x-1.x audio output",
-    "alsa",
-    "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
-    "under development"
-};
-
-LIBAO_EXTERN(alsa)
-
-static snd_pcm_t *alsa_handler;
-static snd_pcm_format_t alsa_format;
-
-#define BUFFER_TIME 500000  // 0.5 s
-#define FRAGCOUNT 16
-
-static size_t bytes_per_sample;
-
-static int alsa_can_pause;
-static snd_pcm_sframes_t prepause_frames;
-
-#define ALSA_DEVICE_SIZE 256
-
-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);
-
-  if (err)
-    mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
-	   file, line, function, tmp, snd_strerror(err));
-  else
-    mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] 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_GET_MUTE:
-  case AOCONTROL_SET_MUTE:
-  case AOCONTROL_GET_VOLUME:
-  case AOCONTROL_SET_VOLUME:
-    {
-      int err;
-      snd_mixer_t *handle;
-      snd_mixer_elem_t *elem;
-      snd_mixer_selem_id_t *sid;
-
-      char *mix_name = "Master";
-      char *card = "default";
-      int mix_index = 0;
-
-      long pmin, pmax;
-      long get_vol, set_vol;
-      float f_multi;
-
-      if(AF_FORMAT_IS_AC3(ao_data.format) || AF_FORMAT_IS_IEC61937(ao_data.format))
-	return CONTROL_TRUE;
-
-      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_tmsg(MSGT_AO,MSGL_ERR,
-		    "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
-		  mix_index = 0 ;
-		}
-	 }
-      }
-      if(mixer_device) card = mixer_device;
-
-      //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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
-	return CONTROL_ERROR;
-      }
-
-      if ((err = snd_mixer_attach(handle, card)) < 0) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
-	snd_mixer_close(handle);
-	return CONTROL_ERROR;
-      }
-      err = snd_mixer_load(handle);
-      if (err < 0) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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));
-
-      switch (cmd) {
-      case AOCONTROL_SET_VOLUME: {
-        ao_control_vol_t *vol = arg;
-	set_vol = vol->left / f_multi + pmin + 0.5;
-
-	//setting channels
-	if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
-		 snd_strerror(err));
-	  goto mixer_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, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
-		 snd_strerror(err));
-	  goto mixer_error;
-	}
-	mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
-	       set_vol, pmin, pmax, f_multi);
-        break;
-      }
-      case AOCONTROL_GET_VOLUME: {
-        ao_control_vol_t *vol = arg;
-        snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
-        vol->left = (get_vol - pmin) * f_multi;
-        snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
-        vol->right = (get_vol - pmin) * f_multi;
-        mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
-        break;
-      }
-      case AOCONTROL_SET_MUTE: {
-        bool *mute = arg;
-	if (!snd_mixer_selem_has_playback_switch(elem))
-          goto mixer_error;
-        if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
-            snd_mixer_selem_set_playback_switch(
-                elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
-	}
-	snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
-                                            !*mute);
-	break;
-      }
-      case AOCONTROL_GET_MUTE: {
-        bool *mute = arg;
-        if (!snd_mixer_selem_has_playback_switch(elem))
-          goto mixer_error;
-        int tmp = 1;
-        snd_mixer_selem_get_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
-                                            &tmp);
-        *mute = !tmp;
-        if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
-            snd_mixer_selem_get_playback_switch(
-                elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
-            *mute &= !tmp;
-        }
-        break;
-      }
-      }
-      snd_mixer_close(handle);
-      return CONTROL_OK;
-    mixer_error:
-      snd_mixer_close(handle);
-      return CONTROL_ERROR;
-    }
-
-  } //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_tmsg (MSGT_AO, MSGL_FATAL,
-    "\n[AO_ALSA] -ao alsa commandline help:\n"\
-    "[AO_ALSA] Example: mpv -ao alsa:device=hw=0.3\n"\
-    "[AO_ALSA]   Sets first card fourth hardware device.\n\n"\
-    "[AO_ALSA] Options:\n"\
-    "[AO_ALSA]   noblock\n"\
-    "[AO_ALSA]     Opens device in non-blocking mode.\n"\
-    "[AO_ALSA]   device=<device-name>\n"\
-    "[AO_ALSA]     Sets device (change , to . and : to =)\n");
-}
-
-static int str_maxlen(void *strp) {
-  strarg_t *str = strp;
-  return str->len <= ALSA_DEVICE_SIZE;
-}
-
-static int try_open_device(const char *device, int open_mode, int try_ac3)
-{
-  int err, len;
-  char *ac3_device, *args;
-
-  if (try_ac3) {
-    /* to set the non-audio bit, use AES0=6 */
-    len = strlen(device);
-    ac3_device = malloc(len + 7 + 1);
-    if (!ac3_device)
-      return -ENOMEM;
-    strcpy(ac3_device, device);
-    args = strchr(ac3_device, ':');
-    if (!args) {
-      /* no existing parameters: add it behind device name */
-      strcat(ac3_device, ":AES0=6");
-    } else {
-      do
-	++args;
-      while (isspace(*args));
-      if (*args == '\0') {
-	/* ":" but no parameters */
-	strcat(ac3_device, "AES0=6");
-      } else if (*args != '{') {
-	/* a simple list of parameters: add it at the end of the list */
-	strcat(ac3_device, ",AES0=6");
-      } else {
-	/* parameters in config syntax: add it inside the { } block */
-	do
-	  --len;
-	while (len > 0 && isspace(ac3_device[len]));
-	if (ac3_device[len] == '}')
-	  strcpy(ac3_device + len, " AES0=6}");
-      }
-    }
-    err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
-		       open_mode);
-    free(ac3_device);
-    if (!err)
-      return 0;
-  }
-  return snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
-                      open_mode);
-}
-
-/*
-    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 chunk_size;
-    snd_pcm_uframes_t bufsize;
-    snd_pcm_uframes_t boundary;
-    const opt_t subopts[] = {
-      {"block", OPT_ARG_BOOL, &block, NULL},
-      {"device", OPT_ARG_STR, &device, 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;
-    mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
-
-    prepause_frames = 0;
-
-    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;
-      case AF_FORMAT_AC3_LE:
-      case AF_FORMAT_S16_LE:
-      case AF_FORMAT_IEC61937_LE:
-	alsa_format = SND_PCM_FORMAT_S16_LE;
-	break;
-      case AF_FORMAT_AC3_BE:
-      case AF_FORMAT_S16_BE:
-      case AF_FORMAT_IEC61937_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_U24_LE:
-	alsa_format = SND_PCM_FORMAT_U24_3LE;
-	break;
-      case AF_FORMAT_U24_BE:
-	alsa_format = SND_PCM_FORMAT_U24_3BE;
-	break;
-      case AF_FORMAT_S24_LE:
-	alsa_format = SND_PCM_FORMAT_S24_3LE;
-	break;
-      case AF_FORMAT_S24_BE:
-	alsa_format = SND_PCM_FORMAT_S24_3BE;
-	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 (AF_FORMAT_IS_AC3(format) || AF_FORMAT_IS_IEC61937(format)) {
-	device.str = "iec958";
-	mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %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;
-	case 8:
-	  if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
-	    device.str = "plug:surround71";
-	  else
-	    device.str = "surround71";
-	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
-	  break;
-	default:
-	  device.str = "default";
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
-        }
-    device.len = strlen(device.str);
-    if (subopt_parse(ao_subdevice, subopts) != 0) {
-        print_help();
-        return 0;
-    }
-    parse_device(alsa_device, device.str, device.len);
-
-    mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
-
-    alsa_can_pause = 1;
-
-    if (!alsa_handler) {
-      int open_mode = block ? 0 : SND_PCM_NONBLOCK;
-      int isac3 =  AF_FORMAT_IS_AC3(format) || AF_FORMAT_IS_IEC61937(format);
-      //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
-      if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
-	{
-	  if (err != -EBUSY && !block) {
-	    mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
-	    if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
-	      mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
-	      return 0;
-	    }
-	  } else {
-	    mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
-	    return 0;
-	  }
-	}
-
-      if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
-         mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
-      } else {
-	mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
-      }
-
-      snd_pcm_hw_params_t *alsa_hwparams;
-      snd_pcm_sw_params_t *alsa_swparams;
-
-      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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_INFO,
-		"[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
-         alsa_format = SND_PCM_FORMAT_S16_LE;
-         if (AF_FORMAT_IS_AC3(ao_data.format))
-           ao_data.format = AF_FORMAT_AC3_LE;
-         else if (AF_FORMAT_IS_IEC61937(ao_data.format))
-           ao_data.format = AF_FORMAT_IEC61937_LE;
-         else
-         ao_data.format = AF_FORMAT_S16_LE;
-      }
-
-      if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
-					      alsa_format)) < 0)
-	{
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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, since that allows users to choose the resampler,
-         even per file if desired */
-      if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
-						     0)) < 0)
-	{
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to disable resampling: %s\n",
-		 snd_strerror(err));
-	  return 0;
-	}
-
-      if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
-						 &ao_data.samplerate, NULL)) < 0)
-        {
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
-		 snd_strerror(err));
-	  return 0;
-        }
-
-      bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
-      bytes_per_sample *= ao_data.channels;
-      ao_data.bps = ao_data.samplerate * bytes_per_sample;
-
-	if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
-							  &(unsigned int){BUFFER_TIME}, NULL)) < 0)
-	  {
-	    mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
-		   snd_strerror(err));
-	    return 0;
-	  }
-
-	if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
-						      &(unsigned int){FRAGCOUNT}, NULL)) < 0) {
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
-		 snd_strerror(err));
-	  return 0;
-	}
-
-      /* finally install hardware parameters */
-      if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
-	{
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set hw-parameters: %s\n",
-		 snd_strerror(err));
-	  return 0;
-	}
-      // end setting hw-params
-
-
-      // gets buffersize for control
-      if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
-	{
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err));
-	  return 0;
-	}
-      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_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err));
-	return 0;
-      } 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
-	       snd_strerror(err));
-	return 0;
-      }
-      if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get boundary: %s\n",
-	       snd_strerror(err));
-	return 0;
-      }
-      /* 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set stop threshold: %s\n",
-	       snd_strerror(err));
-	return 0;
-      }
-      /* play silence when there is an underrun */
-      if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set silence size: %s\n",
-	       snd_strerror(err));
-	return 0;
-      }
-      if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
-	       snd_strerror(err));
-	return 0;
-      }
-      /* end setting sw-params */
-
-      alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
-
-      mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
-	     ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
-	     snd_pcm_format_description(alsa_format));
-
-    } // end switch alsa_handler (spdif)
-    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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err));
-	return;
-      }
-    else {
-      alsa_handler = NULL;
-      mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
-    }
-  }
-  else {
-    mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm pause error: %s\n", snd_strerror(err));
-            return;
-        }
-          mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
-    } else {
-        if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
-            || prepause_frames < 0)
-            prepause_frames = 0;
-
-        if ((err = snd_pcm_drop(alsa_handler)) < 0)
-        {
-            mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err));
-            return;
-        }
-    }
-}
-
-static void audio_resume(void)
-{
-    int err;
-
-    if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
-        mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
-        while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
-    }
-    if (alsa_can_pause) {
-        if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
-        {
-            mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
-            return;
-        }
-        if (prepause_frames) {
-            void *silence = calloc(prepause_frames, bytes_per_sample);
-            play(silence, prepause_frames * bytes_per_sample, 0);
-            free(silence);
-        }
-    }
-}
-
-/* stop playing and empty buffers (for seeking/pause) */
-static void reset(void)
-{
-    int err;
-
-    prepause_frames = 0;
-    if ((err = snd_pcm_drop(alsa_handler)) < 0)
-    {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
-	return;
-    }
-    if ((err = snd_pcm_prepare(alsa_handler)) < 0)
-    {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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 <marius@rospot.com> for giving us the light ;)
-*/
-
-static int play(void* data, int len, int flags)
-{
-  int num_frames;
-  snd_pcm_sframes_t res = 0;
-  if (!(flags & AOPLAY_FINAL_CHUNK))
-      len = len / ao_data.outburst * ao_data.outburst;
-  num_frames = len / bytes_per_sample;
-
-  //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
-
-  if (!alsa_handler) {
-    mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
-	while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
-	  sleep(1);
-      }
-      if (res < 0) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Write error: %s\n", snd_strerror(res));
-	mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Trying to reset soundcard.\n");
-	if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Cannot get pcm status: %s\n", snd_strerror(ret));
-	return 0;
-    }
-
-    unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
-    if (space > ao_data.buffersize) // Buffer underrun?
-        space = ao_data.buffersize;
-    return space;
-}
-
-/* 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 */
-      snd_pcm_forward(alsa_handler, -delay);
-      delay = 0;
-    }
-    return (float)delay / (float)ao_data.samplerate;
-  } else {
-    return 0;
-  }
-}