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, 868 insertions, 0 deletions

diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
new file mode 100644
index 0000000000..27119112cb
--- /dev/null
+++ b/audio/out/ao_alsa.c
@@ -0,0 +1,868 @@
+/*
+ * 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;
+  }
+}