Merge branch 'audio_changes'

Conflicts:
	audio/out/ao_lavc.c

16 files changed, 960 insertions, 947 deletions

diff --git a/audio/out/ao.c b/audio/out/ao.c
index 9fb201a333..10badcfa07 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -250,14 +250,24 @@ void ao_resume(struct ao *ao)
         ao->driver->resume(ao);
 }
 
+bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
+                         struct mp_chmap *map)
+{
+    return mp_chmap_sel_adjust(s, map);
+}
 
+bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
+                          struct mp_chmap *map, int num)
+{
+    return mp_chmap_sel_get_def(s, map, num);
+}
 
 int old_ao_init(struct ao *ao, char *params)
 {
     assert(!global_ao);
     global_ao = ao;
     ao_subdevice = params ? talloc_strdup(ao, params) : NULL;
-    if (ao->driver->old_functions->init(ao->samplerate, ao->channels,
+    if (ao->driver->old_functions->init(ao->samplerate, &ao->channels,
                                         ao->format, 0) == 0) {
         global_ao = NULL;
         return -1;
diff --git a/audio/out/ao.h b/audio/out/ao.h
index 4a7c928824..d908841457 100644
--- a/audio/out/ao.h
+++ b/audio/out/ao.h
@@ -23,6 +23,8 @@
 
 #include "core/bstr.h"
 #include "core/mp_common.h"
+#include "audio/chmap.h"
+#include "audio/chmap_sel.h"
 
 enum aocontrol {
     // _VOLUME commands take struct ao_control_vol pointer for input/output.
@@ -55,7 +57,7 @@ typedef struct ao_info {
 /* interface towards mplayer and */
 typedef struct ao_old_functions {
     int (*control)(int cmd, void *arg);
-    int (*init)(int rate, int channels, int format, int flags);
+    int (*init)(int rate, const struct mp_chmap *channels, int format, int flags);
     void (*uninit)(int immed);
     void (*reset)(void);
     int (*get_space)(void);
@@ -68,7 +70,6 @@ typedef struct ao_old_functions {
 struct ao;
 
 struct ao_driver {
-    bool is_new;
     bool encode;
     const struct ao_info *info;
     const struct ao_old_functions *old_functions;
@@ -86,9 +87,9 @@ struct ao_driver {
 /* global data used by mplayer and plugins */
 struct ao {
     int samplerate;
-    int channels;
+    struct mp_chmap channels;
     int format;
-    int bps;
+    int bps; // bytes per second
     int outburst;
     int buffersize;
     double pts;
@@ -121,6 +122,11 @@ void ao_reset(struct ao *ao);
 void ao_pause(struct ao *ao);
 void ao_resume(struct ao *ao);
 
+bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
+                         struct mp_chmap *map);
+bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
+                          struct mp_chmap *map, int num);
+
 int old_ao_control(struct ao *ao, enum aocontrol cmd, void *arg);
 int old_ao_init(struct ao *ao, char *params);
 void old_ao_uninit(struct ao *ao, bool cut_audio);
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index 366b912b76..93327881e5 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -36,6 +36,7 @@
 #include <alloca.h>
 
 #include "config.h"
+#include "core/options.h"
 #include "core/subopt-helper.h"
 #include "audio/mixer.h"
 #include "core/mp_msg.h"
@@ -46,292 +47,372 @@
 #include <alsa/asoundlib.h>
 
 #include "ao.h"
-#include "audio_out_internal.h"
 #include "audio/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"
+#include "audio/reorder_ch.h"
+
+struct priv {
+    snd_pcm_t *alsa;
+    snd_pcm_format_t alsa_fmt;
+    size_t bytes_per_sample;
+    int can_pause;
+    snd_pcm_sframes_t prepause_frames;
+    float delay_before_pause;
 };
 
-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;
+#define ALSA_DEVICE_SIZE 256
 
-static int alsa_can_pause;
-static snd_pcm_sframes_t prepause_frames;
-static float delay_before_pause;
+#define CHECK_ALSA_ERROR(message) \
+    do { \
+        if (err < 0) { \
+            mp_msg(MSGT_VO, MSGL_ERR, "[AO_ALSA] %s: %s\n", \
+                   (message), snd_strerror(err)); \
+            goto alsa_error; \
+        } \
+    } while (0)
 
-#define ALSA_DEVICE_SIZE 256
+static float get_delay(struct ao *ao);
+static int play(struct ao *ao, void *data, int len, int flags);
 
 static void alsa_error_handler(const char *file, int line, const char *function,
-			       int err, const char *format, ...)
+                               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);
+    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)
+static int control(struct ao *ao, enum aocontrol cmd, void *arg)
 {
-  switch(cmd) {
-  case AOCONTROL_GET_MUTE:
-  case AOCONTROL_SET_MUTE:
-  case AOCONTROL_GET_VOLUME:
-  case AOCONTROL_SET_VOLUME:
+    snd_mixer_t *handle = NULL;
+    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_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;
+        int err;
+        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_IEC61937(ao->format))
+            return CONTROL_TRUE;
+
+        if (ao->opts->mixer_channel) {
+            char *test_mix_index;
+
+            mix_name = strdup(ao->opts->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 (ao->opts->mixer_device)
+            card = ao->opts->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 (ao->opts->mixer_channel) {
+            free(mix_name);
+            mix_name = NULL;
         }
-        break;
-      }
-      }
-      snd_mixer_close(handle);
-      return CONTROL_OK;
-    mixer_error:
-      snd_mixer_close(handle);
-      return CONTROL_ERROR;
+
+        err = snd_mixer_open(&handle, 0);
+        CHECK_ALSA_ERROR("Mixer open error");
+
+        err = snd_mixer_attach(handle, card);
+        CHECK_ALSA_ERROR("Mixer attach error");
+
+        err = snd_mixer_selem_register(handle, NULL, NULL);
+        CHECK_ALSA_ERROR("Mixer register error");
+
+        err = snd_mixer_load(handle);
+        CHECK_ALSA_ERROR("Mixer load 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));
+            goto alsa_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
+            err = snd_mixer_selem_set_playback_volume
+                    (elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol);
+            CHECK_ALSA_ERROR("Error setting left channel");
+            mp_msg(MSGT_AO, MSGL_DBG2, "left=%li, ", set_vol);
+
+            set_vol = vol->right / f_multi + pmin + 0.5;
+
+            err = snd_mixer_selem_set_playback_volume
+                    (elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol);
+            CHECK_ALSA_ERROR("Error setting right channel");
+            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 alsa_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 alsa_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;
     }
 
-  } //end switch
-  return CONTROL_UNKNOWN;
+    } //end switch
+    return CONTROL_UNKNOWN;
+
+alsa_error:
+    if (handle)
+        snd_mixer_close(handle);
+    return CONTROL_ERROR;
 }
 
-static void parse_device (char *dest, const char *src, int len)
+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] = ':';
+    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)
+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");
+    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 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)
+
+static const int mp_to_alsa_format[][2] = {
+    {AF_FORMAT_S8,          SND_PCM_FORMAT_S8},
+    {AF_FORMAT_U8,          SND_PCM_FORMAT_U8},
+    {AF_FORMAT_U16_LE,      SND_PCM_FORMAT_U16_LE},
+    {AF_FORMAT_U16_BE,      SND_PCM_FORMAT_U16_BE},
+    {AF_FORMAT_S16_LE,      SND_PCM_FORMAT_S16_LE},
+    {AF_FORMAT_S16_BE,      SND_PCM_FORMAT_S16_BE},
+    {AF_FORMAT_U32_LE,      SND_PCM_FORMAT_U32_LE},
+    {AF_FORMAT_U32_BE,      SND_PCM_FORMAT_U32_BE},
+    {AF_FORMAT_S32_LE,      SND_PCM_FORMAT_S32_LE},
+    {AF_FORMAT_S32_BE,      SND_PCM_FORMAT_S32_BE},
+    {AF_FORMAT_U24_LE,      SND_PCM_FORMAT_U24_3LE},
+    {AF_FORMAT_U24_BE,      SND_PCM_FORMAT_U24_3BE},
+    {AF_FORMAT_S24_LE,      SND_PCM_FORMAT_S24_3LE},
+    {AF_FORMAT_S24_BE,      SND_PCM_FORMAT_S24_3BE},
+    {AF_FORMAT_FLOAT_LE,    SND_PCM_FORMAT_FLOAT_LE},
+    {AF_FORMAT_FLOAT_BE,    SND_PCM_FORMAT_FLOAT_BE},
+    {AF_FORMAT_AC3_LE,      SND_PCM_FORMAT_S16_LE},
+    {AF_FORMAT_AC3_BE,      SND_PCM_FORMAT_S16_BE},
+    {AF_FORMAT_IEC61937_LE, SND_PCM_FORMAT_S16_LE},
+    {AF_FORMAT_IEC61937_BE, SND_PCM_FORMAT_S16_BE},
+    {AF_FORMAT_MPEG2,       SND_PCM_FORMAT_MPEG},
+    {AF_FORMAT_UNKNOWN,     SND_PCM_FORMAT_UNKNOWN},
+};
+
+static int find_alsa_format(int af_format)
 {
-  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}");
-      }
+    for (int n = 0; mp_to_alsa_format[n][0] != AF_FORMAT_UNKNOWN; n++) {
+        if (mp_to_alsa_format[n][0] == af_format)
+            return mp_to_alsa_format[n][1];
+    }
+    return SND_PCM_FORMAT_UNKNOWN;
+}
+
+// Lists device names and their implied channel map.
+// The second item must be resolvable with mp_chmap_from_str().
+// Source: http://www.alsa-project.org/main/index.php/DeviceNames
+// (Speaker names are slightly different from mpv's.)
+static const char *device_channel_layouts[][2] = {
+    {"default",         "fc"},
+    {"default",         "fl-fr"},
+    {"rear",            "bl-br"},
+    {"center_lfe",      "fc-lfe"},
+    {"side",            "sl-sr"},
+    {"surround40",      "fl-fr-fc-bc"},
+    {"surround50",      "fl-fr-bl-br-fc"},
+    {"surround41",      "fl-fr-bl-br-lfe"},
+    {"surround51",      "fl-fr-bl-br-fc-lfe"},
+    {"surround71",      "fl-fr-bl-br-fc-lfe-sl-sr"},
+};
+
+#define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0]))
+
+#define NUM_ALSA_CHMAPS ARRAY_LEN(device_channel_layouts)
+
+static const char *select_chmap(struct ao *ao)
+{
+    struct mp_chmap_sel sel = {0};
+    struct mp_chmap maps[NUM_ALSA_CHMAPS];
+    for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
+        mp_chmap_from_str(&maps[n], bstr0(device_channel_layouts[n][1]));
+        mp_chmap_sel_add_map(&sel, &maps[n]);
+    };
+
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        return NULL;
+
+    for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
+        if (mp_chmap_equals(&ao->channels, &maps[n]))
+            return device_channel_layouts[n][0];
+    }
+
+    char *name = mp_chmap_to_str(&ao->channels);
+    mp_tmsg(MSGT_AO, MSGL_ERR,
+            "[AO_ALSA] channel layout %s (%d ch) not supported.\n",
+            name, ao->channels.num);
+    talloc_free(name);
+    return "default";
+}
+
+static int try_open_device(struct ao *ao, const char *device, int open_mode,
+                           int try_ac3)
+{
+    struct priv *p = ao->priv;
+    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
+                (&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, open_mode);
+        free(ac3_device);
+        if (!err)
+            return 0;
     }
-    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);
+    return snd_pcm_open
+            (&p->alsa, 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)
+    return: 0=success -1=fail
+ */
+static int init(struct ao *ao, char *params)
 {
     int err;
     int block;
@@ -340,89 +421,29 @@ static int init(int rate_hz, int channels, int format, int flags)
     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}
+        {"block", OPT_ARG_BOOL, &block, NULL},
+        {"device", OPT_ARG_STR, &device, str_maxlen},
+        {NULL}
     };
 
+    struct priv *p = talloc_zero(ao, struct priv);
+    ao->priv = p;
+
     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());
+    mp_msg(MSGT_AO, MSGL_V,
+           "alsa-init: requested format: %d Hz, %d channels, %x\n",
+           ao->samplerate, ao->channels.num, ao->format);
+    p->alsa = NULL;
+    mp_msg(MSGT_AO, MSGL_V, "alsa-init: using ALSA %s\n", snd_asoundlib_version());
 
-    prepause_frames = 0;
-    delay_before_pause = 0;
+    p->prepause_frames = 0;
+    p->delay_before_pause = 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;
-
-      default:
-	alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
-	break;
-      }
-
     //subdevice parsing
     // set defaults
     block = 1;
@@ -432,347 +453,271 @@ static int init(int rate_hz, int channels, int format, int flags)
      * while opening the abstract alias for the spdif subdevice
      * 'iec958'
      */
-    if (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.str = NULL;
+    if (AF_FORMAT_IS_IEC61937(ao->format)) {
+        device.str = "iec958";
+        mp_msg(MSGT_AO, MSGL_V,
+               "alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n",
+               ao->channels.num);
+    } else {
+        device.str = select_chmap(ao);
+        if (strcmp(device.str, "default") != 0 && ao->format == AF_FORMAT_FLOAT_NE)
+        {
+            // hack - use the converter plugin (why the heck?)
+            device.str = talloc_asprintf(ao, "plug:%s", device.str);
         }
+    }
     device.len = strlen(device.str);
-    if (subopt_parse(ao_subdevice, subopts) != 0) {
+    if (subopt_parse(params, 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_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;
+    mp_msg(MSGT_AO, MSGL_V, "alsa-init: using device %s\n", alsa_device);
+
+    p->can_pause = 1;
+
+    int open_mode = block ? 0 : SND_PCM_NONBLOCK;
+    int isac3 =  AF_FORMAT_IS_IEC61937(ao->format);
+    //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
+    err = try_open_device(ao, alsa_device, open_mode, isac3);
+    if (err < 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");
+            err = try_open_device(ao, alsa_device, 0, isac3);
         }
+        CHECK_ALSA_ERROR("Playback open error");
+    }
 
-      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;
+    err = snd_pcm_nonblock(p->alsa, 0);
+    if (err < 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
+    err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
+    CHECK_ALSA_ERROR("Unable to get initial parameters");
+
+    err = snd_pcm_hw_params_set_access
+            (p->alsa, alsa_hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
+    CHECK_ALSA_ERROR("Unable to set access type");
+
+    p->alsa_fmt = find_alsa_format(ao->format);
+    if (p->alsa_fmt == SND_PCM_FORMAT_UNKNOWN) {
+        p->alsa_fmt = SND_PCM_FORMAT_S16;
+        ao->format = AF_FORMAT_S16_NE;
+    }
+
+    err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
+    if (err < 0) {
+        mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Format %s is not supported "
+                "by hardware, trying default.\n", af_fmt2str_short(ao->format));
+        p->alsa_fmt = SND_PCM_FORMAT_S16_LE;
+        if (AF_FORMAT_IS_AC3(ao->format))
+            ao->format = AF_FORMAT_AC3_LE;
+        else if (AF_FORMAT_IS_IEC61937(ao->format))
+            ao->format = AF_FORMAT_IEC61937_LE;
+        else
+            ao->format = AF_FORMAT_S16_LE;
+    }
+
+    err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
+    CHECK_ALSA_ERROR("Unable to set format");
+
+    int num_channels = ao->channels.num;
+    err = snd_pcm_hw_params_set_channels_near
+            (p->alsa, alsa_hwparams, &num_channels);
+    CHECK_ALSA_ERROR("Unable to set channels");
+
+    if (num_channels != ao->channels.num) {
+        mp_tmsg(MSGT_AO, MSGL_ERR,
+                "[AO_ALSA] Couldn't get requested number of channels.\n");
+        mp_chmap_from_channels_alsa(&ao->channels, num_channels);
+    }
+
+    /* 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 */
+    err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
+    CHECK_ALSA_ERROR("Unable to disable resampling");
+
+    err = snd_pcm_hw_params_set_rate_near
+            (p->alsa, alsa_hwparams, &ao->samplerate, NULL);
+    CHECK_ALSA_ERROR("Unable to set samplerate-2");
+
+    p->bytes_per_sample = af_fmt2bits(ao->format) / 8;
+    p->bytes_per_sample *= ao->channels.num;
+    ao->bps = ao->samplerate * p->bytes_per_sample;
+
+    err = snd_pcm_hw_params_set_buffer_time_near
+            (p->alsa, alsa_hwparams, &(unsigned int){BUFFER_TIME}, NULL);
+    CHECK_ALSA_ERROR("Unable to set buffer time near");
+
+    err = snd_pcm_hw_params_set_periods_near
+            (p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
+    CHECK_ALSA_ERROR("Unable to set periods");
+
+    /* finally install hardware parameters */
+    err = snd_pcm_hw_params(p->alsa, alsa_hwparams);
+    CHECK_ALSA_ERROR("Unable to set hw-parameters");
+
+    // end setting hw-params
+
+    // gets buffersize for control
+    err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize);
+    CHECK_ALSA_ERROR("Unable to get buffersize");
+
+    ao->buffersize = bufsize * p->bytes_per_sample;
+    mp_msg(MSGT_AO, MSGL_V, "alsa-init: got buffersize=%i\n",
+            ao->buffersize);
+
+    err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL);
+    CHECK_ALSA_ERROR("Unable to get period size");
+
+    mp_msg(MSGT_AO, MSGL_V, "alsa-init: got period size %li\n", chunk_size);
+    ao->outburst = chunk_size * p->bytes_per_sample;
+
+    /* setting software parameters */
+    err = snd_pcm_sw_params_current(p->alsa, alsa_swparams);
+    CHECK_ALSA_ERROR("Unable to get sw-parameters");
+
+    err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary);
+    CHECK_ALSA_ERROR("Unable to get boundary");
+
+    /* start playing when one period has been written */
+    err = snd_pcm_sw_params_set_start_threshold
+            (p->alsa, alsa_swparams, chunk_size);
+    CHECK_ALSA_ERROR("Unable to set start threshold");
+
+    /* disable underrun reporting */
+    err = snd_pcm_sw_params_set_stop_threshold
+            (p->alsa, alsa_swparams, boundary);
+    CHECK_ALSA_ERROR("Unable to set stop threshold");
+
+    /* play silence when there is an underrun */
+    err = snd_pcm_sw_params_set_silence_size
+            (p->alsa, alsa_swparams, boundary);
+    CHECK_ALSA_ERROR("Unable to set silence size");
+
+    err = snd_pcm_sw_params(p->alsa, alsa_swparams);
+    CHECK_ALSA_ERROR("Unable to get sw-parameters");
+
+    /* end setting sw-params */
+
+    p->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->samplerate, ao->channels.num, (int)p->bytes_per_sample,
+            ao->buffersize, snd_pcm_format_description(p->alsa_fmt));
+
+    return 0;
+
+alsa_error:
+    return -1;
 } // end init
 
 
 /* close audio device */
-static void uninit(int immed)
+static void uninit(struct ao *ao, bool immed)
 {
+    struct priv *p = ao->priv;
 
-  if (alsa_handler) {
-    int err;
+    if (p->alsa) {
+        int err;
+
+        if (!immed)
+            snd_pcm_drain(p->alsa);
 
-    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");
+        err = snd_pcm_close(p->alsa);
+        CHECK_ALSA_ERROR("pcm close error");
+
+        p->alsa = 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");
     }
-  }
-  else {
-    mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n");
-  }
+
+alsa_error: ;
 }
 
-static void audio_pause(void)
+static void audio_pause(struct ao *ao)
 {
+    struct priv *p = ao->priv;
     int err;
 
-    if (alsa_can_pause) {
-        delay_before_pause = get_delay();
-        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");
+    if (p->can_pause) {
+        p->delay_before_pause = get_delay(ao);
+        err = snd_pcm_pause(p->alsa, 1);
+        CHECK_ALSA_ERROR("pcm pause error");
+        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;
-        delay_before_pause = prepause_frames / (float)ao_data.samplerate;
+        if (snd_pcm_delay(p->alsa, &p->prepause_frames) < 0
+            || p->prepause_frames < 0)
+            p->prepause_frames = 0;
+        p->delay_before_pause = p->prepause_frames / (float)ao->samplerate;
 
-        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;
-        }
+        err = snd_pcm_drop(p->alsa);
+        CHECK_ALSA_ERROR("pcm drop error");
     }
+
+alsa_error: ;
 }
 
-static void audio_resume(void)
+static void audio_resume(struct ao *ao)
 {
+    struct priv *p = ao->priv;
     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 (snd_pcm_state(p->alsa) == 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(p->alsa)) == -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");
+    if (p->can_pause) {
+        err = snd_pcm_pause(p->alsa, 0);
+        CHECK_ALSA_ERROR("pcm resume error");
+        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);
+        err = snd_pcm_prepare(p->alsa);
+        CHECK_ALSA_ERROR("pcm prepare error");
+        if (p->prepause_frames) {
+            void *silence = calloc(p->prepause_frames, p->bytes_per_sample);
+            play(ao, silence, p->prepause_frames * p->bytes_per_sample, 0);
             free(silence);
         }
     }
+
+alsa_error: ;
 }
 
 /* stop playing and empty buffers (for seeking/pause) */
-static void reset(void)
+static void reset(struct ao *ao)
 {
+    struct priv *p = ao->priv;
     int err;
 
-    prepause_frames = 0;
-    delay_before_pause = 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;
+    p->prepause_frames = 0;
+    p->delay_before_pause = 0;
+    err = snd_pcm_drop(p->alsa);
+    CHECK_ALSA_ERROR("pcm prepare error");
+    err = snd_pcm_prepare(p->alsa);
+    CHECK_ALSA_ERROR("pcm prepare error");
+
+alsa_error: ;
 }
 
 /*
@@ -780,91 +725,115 @@ static void reset(void)
     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)
+static int play(struct ao *ao, 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;
+    struct priv *p = ao->priv;
+    int num_frames;
+    snd_pcm_sframes_t res = 0;
+    if (!(flags & AOPLAY_FINAL_CHUNK))
+        len = len / ao->outburst * ao->outburst;
+    num_frames = len / p->bytes_per_sample;
 
-  //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
+    //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 (!p->alsa) {
+        mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Device configuration error.");
+        return 0;
+    }
 
-  if (num_frames == 0)
-    return 0;
+    if (num_frames == 0)
+        return 0;
+
+    do {
+        res = snd_pcm_writei(p->alsa, 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(p->alsa)) == -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");
+            res = snd_pcm_prepare(p->alsa);
+            int err = res;
+            CHECK_ALSA_ERROR("pcm prepare error");
+            res = 0;
+        }
+    } while (res == 0);
+
+    return res < 0 ? 0 : res * p->bytes_per_sample;
 
-  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));
-	  break;
-	}
-	res = 0;
-      }
-  } while (res == 0);
-
-  return res < 0 ? 0 : res * bytes_per_sample;
+alsa_error:
+    return 0;
 }
 
 /* how many byes are free in the buffer */
-static int get_space(void)
+static int get_space(struct ao *ao)
 {
+    struct priv *p = ao->priv;
     snd_pcm_status_t *status;
-    int ret;
+    int err;
 
     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;
-    }
+    err = snd_pcm_status(p->alsa, status);
+    CHECK_ALSA_ERROR("cannot get pcm status");
 
-    unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
-    if (space > ao_data.buffersize) // Buffer underrun?
-        space = ao_data.buffersize;
+    unsigned space = snd_pcm_status_get_avail(status) * p->bytes_per_sample;
+    if (space > ao->buffersize) // Buffer underrun?
+        space = ao->buffersize;
     return space;
+
+alsa_error:
+    return 0;
 }
 
 /* delay in seconds between first and last sample in buffer */
-static float get_delay(void)
+static float get_delay(struct ao *ao)
 {
-  if (alsa_handler) {
-    snd_pcm_sframes_t delay;
+    struct priv *p = ao->priv;
+    if (p->alsa) {
+        snd_pcm_sframes_t delay;
 
-    if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_PAUSED)
-      return delay_before_pause;
+        if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED)
+            return p->delay_before_pause;
 
-    if (snd_pcm_delay(alsa_handler, &delay) < 0)
-      return 0;
+        if (snd_pcm_delay(p->alsa, &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;
-  }
+        if (delay < 0) {
+            /* underrun - move the application pointer forward to catch up */
+            snd_pcm_forward(p->alsa, -delay);
+            delay = 0;
+        }
+        return (float)delay / (float)ao->samplerate;
+    } else
+        return 0;
 }
+
+const struct ao_driver audio_out_alsa = {
+    .info = &(const struct ao_info) {
+        "ALSA-0.9.x-1.x audio output",
+        "alsa",
+        "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
+        "under development"
+    },
+    .init      = init,
+    .uninit    = uninit,
+    .control   = control,
+    .get_space = get_space,
+    .play      = play,
+    .get_delay = get_delay,
+    .pause     = audio_pause,
+    .resume    = audio_resume,
+    .reset     = reset,
+};
diff --git a/audio/out/ao_coreaudio.c b/audio/out/ao_coreaudio.c
index ba8f517c75..54ff1b4915 100644
--- a/audio/out/ao_coreaudio.c
+++ b/audio/out/ao_coreaudio.c
@@ -412,7 +412,7 @@ static void print_help(void)
     free(devids);
 }
 
-static int init(int rate,int channels,int format,int flags)
+static int init(int rate,const struct mp_chmap *channels,int format,int flags)
 {
 AudioStreamBasicDescription inDesc;
 AudioComponentDescription desc;
@@ -439,7 +439,7 @@ int device_id, display_help = 0;
             return 0;
     }
 
-    ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
+    ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, ao_data.channels.num, af_fmt2str_short(format), flags);
 
     ao = calloc(1, sizeof(ao_coreaudio_t));
 
@@ -499,10 +499,15 @@ int device_id, display_help = 0;
     // Save selected device id
     ao->i_selected_dev = devid_def;
 
+    struct mp_chmap_sel chmap_sel = {0};
+    mp_chmap_sel_add_waveext(&chmap_sel);
+    if (!ao_chmap_sel_adjust(&ao_data, &ao_data.channels, &chmap_sel))
+        goto err_out;
+
 	// Build Description for the input format
 	inDesc.mSampleRate=rate;
 	inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
-	inDesc.mChannelsPerFrame=channels;
+	inDesc.mChannelsPerFrame=ao_data.channels.num;
 	inDesc.mBitsPerChannel=af_fmt2bits(format);
 
     if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
@@ -521,7 +526,7 @@ int device_id, display_help = 0;
         inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
 
     inDesc.mFramesPerPacket = 1;
-    ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
+    ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*ao_data.channels.num*(inDesc.mBitsPerChannel/8);
     print_format(MSGL_V, "source:",&inDesc);
 
     if (ao->b_supports_digital)
@@ -605,7 +610,8 @@ int device_id, display_help = 0;
 	ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
 
 	ao_data.samplerate = inDesc.mSampleRate;
-	ao_data.channels = inDesc.mChannelsPerFrame;
+        if (!ao_chmap_sel_get_def(&ao_data, &chmap_sel, &ao_data.channels, inDesc.mChannelsPerFrame))
+            goto err_out2;
     ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
     ao_data.outburst = ao->chunk_size;
 	ao_data.buffersize = ao_data.bps;
@@ -837,7 +843,8 @@ static int OpenSPDIF(void)
     ao->chunk_size = ao->stream_format.mBytesPerPacket;
 
     ao_data.samplerate = ao->stream_format.mSampleRate;
-    ao_data.channels = ao->stream_format.mChannelsPerFrame;
+    // Applies default ordering; ok because AC3 data is always in mpv internal channel order
+    mp_chmap_from_channels(&ao_data.channels, ao->stream_format.mChannelsPerFrame);
     ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
     ao_data.outburst = ao->chunk_size;
     ao_data.buffersize = ao_data.bps;
diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index 9ad27306bd..967084d3ae 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -64,26 +64,6 @@ LIBAO_EXTERN(dsound)
 
 static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}};
 
-#define SPEAKER_FRONT_LEFT             0x1
-#define SPEAKER_FRONT_RIGHT            0x2
-#define SPEAKER_FRONT_CENTER           0x4
-#define SPEAKER_LOW_FREQUENCY          0x8
-#define SPEAKER_BACK_LEFT              0x10
-#define SPEAKER_BACK_RIGHT             0x20
-#define SPEAKER_FRONT_LEFT_OF_CENTER   0x40
-#define SPEAKER_FRONT_RIGHT_OF_CENTER  0x80
-#define SPEAKER_BACK_CENTER            0x100
-#define SPEAKER_SIDE_LEFT              0x200
-#define SPEAKER_SIDE_RIGHT             0x400
-#define SPEAKER_TOP_CENTER             0x800
-#define SPEAKER_TOP_FRONT_LEFT         0x1000
-#define SPEAKER_TOP_FRONT_CENTER       0x2000
-#define SPEAKER_TOP_FRONT_RIGHT        0x4000
-#define SPEAKER_TOP_BACK_LEFT          0x8000
-#define SPEAKER_TOP_BACK_CENTER        0x10000
-#define SPEAKER_TOP_BACK_RIGHT         0x20000
-#define SPEAKER_RESERVED               0x80000000
-
 #if 0
 #define DSSPEAKER_HEADPHONE         0x00000001
 #define DSSPEAKER_MONO              0x00000002
@@ -107,28 +87,6 @@ typedef struct {
 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
 #endif
 
-static const int channel_mask[] = {
-  /*  1 */ SPEAKER_FRONT_CENTER,
-  /*  2 */ SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT,
-  /*  3 */ SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT
-         | SPEAKER_LOW_FREQUENCY,
-  /*  4 */ SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT
-         | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT,
-  /*  5 */ SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT
-         | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT
-         | SPEAKER_LOW_FREQUENCY,
-  /*  6 */ SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT
-         | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT
-         | SPEAKER_LOW_FREQUENCY,
-  /*  7 */ SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT
-         | SPEAKER_BACK_LEFT    | SPEAKER_BACK_CENTER  | SPEAKER_BACK_RIGHT
-         | SPEAKER_LOW_FREQUENCY,
-  /*  8 */ SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT
-         | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT
-         | SPEAKER_SIDE_LEFT    | SPEAKER_SIDE_RIGHT
-         | SPEAKER_LOW_FREQUENCY,
-};
-
 static HINSTANCE hdsound_dll = NULL;      ///handle to the dll
 static LPDIRECTSOUND hds = NULL;          ///direct sound object
 static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer
@@ -349,22 +307,9 @@ static int write_buffer(unsigned char *data, int len)
   if (SUCCEEDED(res))
   {
   	if (!AF_FORMAT_IS_AC3(ao_data.format)) {
-            int sampsize = af_fmt2bits(ao_data.format) / 8;
-            reorder_channel_copy_nch(data,
-                                     AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
-                                     lpvPtr1,
-                                     AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
-                                     ao_data.channels,
-                                     dwBytes1 / sampsize,
-                                     sampsize);
+            memcpy(lpvPtr1, data, dwBytes1);
             if (lpvPtr2 != NULL)
-                reorder_channel_copy_nch(data + dwBytes1,
-                                         AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
-                                         lpvPtr2,
-                                         AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
-                                         ao_data.channels,
-                                         dwBytes2 / sampsize,
-                                         sampsize);
+                memcpy(lpvPtr2, (char *)data + dwBytes1, dwBytes2);
 
   	    write_offset+=dwBytes1+dwBytes2;
   	    if(write_offset>=buffer_size)
@@ -432,7 +377,7 @@ static int control(int cmd, void *arg)
 \param flags unused
 \return 1=success 0=fail
 */
-static int init(int rate, int channels, int format, int flags)
+static int init(int rate, const struct mp_chmap *channels, int format, int flags)
 {
     int res;
 	if (!InitDirectSound()) return 0;
@@ -445,15 +390,14 @@ static int init(int rate, int channels, int format, int flags)
 	DSBUFFERDESC dsbpridesc;
 	DSBUFFERDESC dsbdesc;
 
-	//check if the channel count and format is supported in general
-	if (channels > FF_ARRAY_ELEMS(channel_mask)) {
-		UninitDirectSound();
-		mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: 8 channel audio not yet supported\n");
-		return 0;
-	}
-
-	if (AF_FORMAT_IS_AC3(format))
+	if (AF_FORMAT_IS_AC3(format)) {
 		format = AF_FORMAT_AC3_NE;
+        } else {
+                struct mp_chmap_sel sel = {0};
+                mp_chmap_sel_add_waveext(&sel);
+                if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
+                        return 0;
+        }
 	switch(format){
 		case AF_FORMAT_AC3_NE:
 		case AF_FORMAT_S24_LE:
@@ -465,25 +409,24 @@ static int init(int rate, int channels, int format, int flags)
 			format=AF_FORMAT_S16_LE;
 	}
 	//fill global ao_data
-	ao_data.channels = channels;
 	ao_data.samplerate = rate;
 	ao_data.format = format;
-	ao_data.bps = channels * rate * (af_fmt2bits(format)>>3);
+	ao_data.bps = ao_data.channels.num * rate * (af_fmt2bits(format)>>3);
 	if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec
-	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, channels, af_fmt2str_short(format));
+	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, ao_data.channels.num, af_fmt2str_short(format));
 	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);
 
 	//fill waveformatex
 	ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
-	wformat.Format.cbSize          = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
-	wformat.Format.nChannels       = channels;
+	wformat.Format.cbSize          = (ao_data.channels.num > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
+	wformat.Format.nChannels       = ao_data.channels.num;
 	wformat.Format.nSamplesPerSec  = rate;
 	if (AF_FORMAT_IS_AC3(format)) {
 		wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
 		wformat.Format.wBitsPerSample  = 16;
 		wformat.Format.nBlockAlign     = 4;
 	} else {
-		wformat.Format.wFormatTag      = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
+		wformat.Format.wFormatTag      = (ao_data.channels.num > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
 		wformat.Format.wBitsPerSample  = af_fmt2bits(format);
 		wformat.Format.nBlockAlign     = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
 	}
@@ -503,8 +446,8 @@ static int init(int rate, int channels, int format, int flags)
 	                | DSBCAPS_GLOBALFOCUS         /** Allows background playing */
 	                | DSBCAPS_CTRLVOLUME;         /** volume control enabled */
 
-	if (channels > 2) {
-		wformat.dwChannelMask = channel_mask[channels - 1];
+	if (ao_data.channels.num > 2) {
+		wformat.dwChannelMask = mp_chmap_to_waveext(&ao_data.channels);
 		wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
 		wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
 		// Needed for 5.1 on emu101k - shit soundblaster
diff --git a/audio/out/ao_jack.c b/audio/out/ao_jack.c
index b75122dc17..0f8baeab86 100644
--- a/audio/out/ao_jack.c
+++ b/audio/out/ao_jack.c
@@ -50,7 +50,7 @@ static const ao_info_t info =
 LIBAO_EXTERN(jack)
 
 //! maximum number of channels supported, avoids lots of mallocs
-#define MAX_CHANS 8
+#define MAX_CHANS MP_NUM_CHANNELS
 static jack_port_t *ports[MAX_CHANS];
 static int num_ports; ///< Number of used ports == number of channels
 static jack_client_t *client;
@@ -202,7 +202,8 @@ static void print_help (void)
          );
 }
 
-static int init(int rate, int channels, int format, int flags) {
+static int init(int rate, const struct mp_chmap *channels, int format, int flags)
+{
   const char **matching_ports = NULL;
   char *port_name = NULL;
   char *client_name = NULL;
@@ -222,10 +223,12 @@ static int init(int rate, int channels, int format, int flags) {
     print_help();
     return 0;
   }
-  if (channels > MAX_CHANS) {
-    mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] Invalid number of channels: %i\n", channels);
+
+  struct mp_chmap_sel sel = {0};
+  mp_chmap_sel_add_waveext(&sel);
+  if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
     goto err_out;
-  }
+
   if (!client_name) {
     client_name = malloc(40);
     sprintf(client_name, "mpv [%d]", getpid());
@@ -249,9 +252,9 @@ static int init(int rate, int channels, int format, int flags) {
     goto err_out;
   }
   i = 1;
+  num_ports = ao_data.channels.num;
   while (matching_ports[i]) i++;
-  if (channels > i) channels = i;
-  num_ports = channels;
+  if (num_ports > i) num_ports = i;
 
   // create out output ports
   for (i = 0; i < num_ports; i++) {
@@ -281,10 +284,12 @@ static int init(int rate, int channels, int format, int flags) {
                  / (float)rate;
   callback_interval = 0;
 
-  ao_data.channels = channels;
+  if (!ao_chmap_sel_get_def(&ao_data, &sel, &ao_data.channels, num_ports))
+    goto err_out;
+
   ao_data.samplerate = rate;
   ao_data.format = AF_FORMAT_FLOAT_NE;
-  ao_data.bps = channels * rate * sizeof(float);
+  ao_data.bps = ao_data.channels.num * rate * sizeof(float);
   ao_data.buffersize = CHUNK_SIZE * NUM_CHUNKS;
   ao_data.outburst = CHUNK_SIZE;
   free(matching_ports);
diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index 33139ce866..d8185a8dca 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -101,7 +101,14 @@ static int init(struct ao *ao, char *params)
     ac->stream->codec->time_base.den = ao->samplerate;
 
     ac->stream->codec->sample_rate = ao->samplerate;
-    ac->stream->codec->channels = ao->channels;
+
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_any(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        return -1;
+    mp_chmap_reorder_to_lavc(&ao->channels);
+    ac->stream->codec->channels = ao->channels.num;
+    ac->stream->codec->channel_layout = mp_chmap_to_lavc(&ao->channels);
 
     ac->stream->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
 
@@ -243,36 +250,6 @@ out_takefirst:
 
     ac->stream->codec->bits_per_raw_sample = ac->sample_size * 8;
 
-    switch (ao->channels) {
-    case 1:
-        ac->stream->codec->channel_layout = AV_CH_LAYOUT_MONO;
-        break;
-    case 2:
-        ac->stream->codec->channel_layout = AV_CH_LAYOUT_STEREO;
-        break;
-    /* someone please check if these are what mplayer normally assumes
-       case 3:
-        ac->stream->codec->channel_layout = AV_CH_LAYOUT_SURROUND;
-        break;
-       case 4:
-        ac->stream->codec->channel_layout = AV_CH_LAYOUT_2_2;
-        break;
-     */
-    case 5:
-        ac->stream->codec->channel_layout = AV_CH_LAYOUT_5POINT0;
-        break;
-    case 6:
-        ac->stream->codec->channel_layout = AV_CH_LAYOUT_5POINT1;
-        break;
-    case 8:
-        ac->stream->codec->channel_layout = AV_CH_LAYOUT_7POINT1;
-        break;
-    default:
-        mp_msg(MSGT_ENCODE, MSGL_ERR,
-               "ao-lavc: unknown channel layout; hoping for the best\n");
-        break;
-    }
-
     if (encode_lavc_open_codec(ao->encode_lavc_ctx, ac->stream) < 0)
         return -1;
 
@@ -282,11 +259,12 @@ out_takefirst:
 
     if (ac->pcmhack) {
         ac->aframesize = 16384; // "enough"
-        ac->buffer_size = ac->aframesize * ac->pcmhack * ao->channels * 2 + 200;
+        ac->buffer_size =
+            ac->aframesize * ac->pcmhack * ao->channels.num * 2 + 200;
     } else {
         ac->aframesize = ac->stream->codec->frame_size;
-        ac->buffer_size = ac->aframesize * ac->sample_size * ao->channels * 2 +
-                          200;
+        ac->buffer_size =
+            ac->aframesize * ac->sample_size * ao->channels.num * 2 + 200;
     }
     if (ac->buffer_size < FF_MIN_BUFFER_SIZE)
         ac->buffer_size = FF_MIN_BUFFER_SIZE;
@@ -304,10 +282,10 @@ out_takefirst:
     ac->offset_left = ac->offset;
 
     //fill_ao_data:
-    ao->outburst = ac->aframesize * ac->sample_size * ao->channels *
-                   ac->framecount;
+    ao->outburst =
+        ac->aframesize * ac->sample_size * ao->channels.num * ac->framecount;
     ao->buffersize = ao->outburst * 2;
-    ao->bps = ao->channels * ao->samplerate * ac->sample_size;
+    ao->bps = ao->channels.num * ao->samplerate * ac->sample_size;
     ao->untimed = true;
     ao->priv = ac;
 
@@ -348,7 +326,7 @@ static void uninit(struct ao *ao, bool cut_audio)
             // TRICK: append aframesize-1 samples to the end, then play() will
             // encode all it can
             size_t extralen =
-                (ac->aframesize - 1) * ao->channels * ac->sample_size;
+                (ac->aframesize - 1) * ao->channels.num * ac->sample_size;
             void *paddingbuf = talloc_size(ao, ao->buffer.len + extralen);
             memcpy(paddingbuf, ao->buffer.start, ao->buffer.len);
             fill_with_padding((char *) paddingbuf + ao->buffer.len,
@@ -392,12 +370,6 @@ static int encode(struct ao *ao, double apts, void *data)
     int status, gotpacket;
 
     ac->aframecount++;
-    if (data && (ao->channels == 5 || ao->channels == 6 || ao->channels == 8)) {
-        reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
-                            AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
-                            ao->channels,
-                            ac->aframesize * ao->channels, ac->sample_size);
-    }
 
     if (data)
         ectx->audio_pts_offset = realapts - apts;
@@ -411,12 +383,18 @@ static int encode(struct ao *ao, double apts, void *data)
         frame->nb_samples = ac->aframesize;
 
         if (ac->planarize) {
-            void *data2 = talloc_size(ao, ac->aframesize * ao->channels * ac->sample_size);
-            reorder_to_planar(data2, data, ac->sample_size, ao->channels, ac->aframesize);
+            void *data2 = talloc_size(ao, ac->aframesize * ao->channels.num *
+                                      ac->sample_size);
+            reorder_to_planar(data2, data, ac->sample_size, ao->channels.num,
+                              ac->aframesize);
             data = data2;
         }
 
-        if (avcodec_fill_audio_frame(frame, ao->channels, ac->stream->codec->sample_fmt, data, ac->aframesize * ao->channels * ac->sample_size, 1)) {
+        size_t audiolen = ac->aframesize * ao->channels.num * ac->sample_size;
+        if (avcodec_fill_audio_frame(frame, ao->channels.num,
+                                     ac->stream->codec->sample_fmt, data,
+                                     audiolen, 1))
+        {
             mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: error filling\n");
             return -1;
         }
@@ -523,7 +501,7 @@ static int play(struct ao *ao, void *data, int len, int flags)
     double pts = ao->pts;
     double outpts;
 
-    len /= ac->sample_size * ao->channels;
+    len /= ac->sample_size * ao->channels.num;
 
     if (!encode_lavc_start(ectx)) {
         mp_msg(MSGT_ENCODE, MSGL_WARN,
@@ -593,7 +571,7 @@ static int play(struct ao *ao, void *data, int len, int flags)
             if (ac->offset_left <= -len) {
                 // skip whole frame
                 ac->offset_left += len;
-                return len * ac->sample_size * ao->channels;
+                return len * ac->sample_size * ao->channels.num;
             } else {
                 // skip part of this frame, buffer/encode the rest
                 bufpos -= ac->offset_left;
@@ -604,11 +582,11 @@ static int play(struct ao *ao, void *data, int len, int flags)
             // make a temporary buffer, filled with zeroes at the start
             // (don't worry, only happens once)
 
-            paddingbuf = talloc_size(ac, ac->sample_size * ao->channels *
+            paddingbuf = talloc_size(ac, ac->sample_size * ao->channels.num *
                                          (ac->offset_left + len));
             fill_with_padding(paddingbuf, ac->offset_left, ac->sample_size,
                               ac->sample_padding);
-            data = (char *) paddingbuf + ac->sample_size * ao->channels *
+            data = (char *) paddingbuf + ac->sample_size * ao->channels.num *
                                          ac->offset_left;
             bufpos -= ac->offset_left; // yes, negative!
             ptsoffset += ac->offset_left;
@@ -651,7 +629,7 @@ static int play(struct ao *ao, void *data, int len, int flags)
     while (len - bufpos >= ac->aframesize) {
         encode(ao,
                outpts + (bufpos + ptsoffset) / (double) ao->samplerate + encode_lavc_getoffset(ectx, ac->stream),
-               (char *) data + ac->sample_size * bufpos * ao->channels);
+               (char *) data + ac->sample_size * bufpos * ao->channels.num);
         bufpos += ac->aframesize;
     }
 
@@ -667,11 +645,10 @@ static int play(struct ao *ao, void *data, int len, int flags)
             ectx->next_in_pts = nextpts;
     }
 
-    return bufpos * ac->sample_size * ao->channels;
+    return bufpos * ac->sample_size * ao->channels.num;
 }
 
 const struct ao_driver audio_out_lavc = {
-    .is_new = true,
     .encode = true,
     .info = &(const struct ao_info) {
         "audio encoding using libavcodec",
diff --git a/audio/out/ao_null.c b/audio/out/ao_null.c
index 102f0a7013..53ec2a9a83 100644
--- a/audio/out/ao_null.c
+++ b/audio/out/ao_null.c
@@ -48,11 +48,17 @@ static int init(struct ao *ao, char *params)
 {
     struct priv *priv = talloc_zero(ao, struct priv);
     ao->priv = priv;
+
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_any(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        return -1;
+
     int samplesize = af_fmt2bits(ao->format) / 8;
-    ao->outburst = 256 * ao->channels * samplesize;
+    ao->outburst = 256 * ao->channels.num * samplesize;
     // A "buffer" for about 0.2 seconds of audio
     ao->buffersize = (int)(ao->samplerate * 0.2 / 256 + 1) * ao->outburst;
-    ao->bps = ao->channels * ao->samplerate * samplesize;
+    ao->bps = ao->channels.num * ao->samplerate * samplesize;
     priv->last_time = GetTimer();
 
     return 0;
@@ -110,7 +116,6 @@ static float get_delay(struct ao *ao)
 }
 
 const struct ao_driver audio_out_null = {
-    .is_new = true,
     .info = &(const struct ao_info) {
         "Null audio output",
         "null",
diff --git a/audio/out/ao_openal.c b/audio/out/ao_openal.c
index 157cf93ac4..08f8bc1978 100644
--- a/audio/out/ao_openal.c
+++ b/audio/out/ao_openal.c
@@ -53,7 +53,7 @@ static const ao_info_t info =
 
 LIBAO_EXTERN(openal)
 
-#define MAX_CHANS 8
+#define MAX_CHANS MP_NUM_CHANNELS
 #define NUM_BUF 128
 #define CHUNK_SIZE 512
 static ALuint buffers[MAX_CHANS][NUM_BUF];
@@ -109,15 +109,29 @@ static void list_devices(void) {
   }
 }
 
-static int init(int rate, int channels, int format, int flags) {
+struct speaker {
+    int id;
+    float pos[3];
+};
+
+static const struct speaker speaker_pos[] = {
+    {MP_SPEAKER_ID_FL,   {-1, 0, 0.5}},
+    {MP_SPEAKER_ID_FR,   { 1, 0, 0.5}},
+    {MP_SPEAKER_ID_FC,   { 0, 0,   1}},
+    {MP_SPEAKER_ID_LFE,  { 0, 0, 0.1}},
+    {MP_SPEAKER_ID_BL,   {-1, 0,  -1}},
+    {MP_SPEAKER_ID_BR,   { 1, 0,  -1}},
+    {MP_SPEAKER_ID_BC,   { 0, 0,  -1}},
+    {MP_SPEAKER_ID_SL,   {-1, 0,   0}},
+    {MP_SPEAKER_ID_SR,   { 1, 0,   0}},
+    {-1},
+};
+
+static int init(int rate, const struct mp_chmap *channels, int format,
+                int flags)
+{
   float position[3] = {0, 0, 0};
   float direction[6] = {0, 0, 1, 0, -1, 0};
-  float sppos[MAX_CHANS][3] = {
-    {-1, 0, 0.5}, {1, 0, 0.5},
-    {-1, 0,  -1}, {1, 0,  -1},
-    {0,  0,   1}, {0, 0, 0.1},
-    {-1, 0,   0}, {1, 0,   0},
-  };
   ALCdevice *dev = NULL;
   ALCcontext *ctx = NULL;
   ALCint freq = 0;
@@ -137,9 +151,22 @@ static int init(int rate, int channels, int format, int flags) {
     list_devices();
     goto err_out;
   }
-  if (channels > MAX_CHANS) {
-    mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Invalid number of channels: %i\n", channels);
+  struct mp_chmap_sel sel = {0};
+  for (i = 0; speaker_pos[i].id != -1; i++)
+    mp_chmap_sel_add_speaker(&sel, speaker_pos[i].id);
+  if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
     goto err_out;
+  struct speaker speakers[MAX_CHANS];
+  for (i = 0; i < ao_data.channels.num; i++) {
+    speakers[i].id = -1;
+    for (int n = 0; speaker_pos[n].id >= 0; n++) {
+      if (speaker_pos[n].id == ao_data.channels.speaker[i])
+        speakers[i] = speaker_pos[n];
+    }
+    if (speakers[i].id < 0) {
+      mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Unknown channel layout\n");
+      goto err_out;
+    }
   }
   dev = alcOpenDevice(device);
   if (!dev) {
@@ -150,25 +177,22 @@ static int init(int rate, int channels, int format, int flags) {
   alcMakeContextCurrent(ctx);
   alListenerfv(AL_POSITION, position);
   alListenerfv(AL_ORIENTATION, direction);
-  alGenSources(channels, sources);
-  for (i = 0; i < channels; i++) {
+  alGenSources(ao_data.channels.num, sources);
+  for (i = 0; i < ao_data.channels.num; i++) {
     cur_buf[i] = 0;
     unqueue_buf[i] = 0;
     alGenBuffers(NUM_BUF, buffers[i]);
-    alSourcefv(sources[i], AL_POSITION, sppos[i]);
+    alSourcefv(sources[i], AL_POSITION, speakers[i].pos);
     alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
   }
-  if (channels == 1)
-    alSource3f(sources[0], AL_POSITION, 0, 0, 1);
-  ao_data.channels = channels;
   alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
   if (alcGetError(dev) == ALC_NO_ERROR && freq)
     rate = freq;
   ao_data.samplerate = rate;
   ao_data.format = AF_FORMAT_S16_NE;
-  ao_data.bps = channels * rate * 2;
+  ao_data.bps = ao_data.channels.num * rate * 2;
   ao_data.buffersize = CHUNK_SIZE * NUM_BUF;
-  ao_data.outburst = channels * CHUNK_SIZE;
+  ao_data.outburst = ao_data.channels.num * CHUNK_SIZE;
   tmpbuf = malloc(CHUNK_SIZE);
   free(device);
   return 1;
@@ -200,7 +224,7 @@ static void uninit(int immed) {
 static void unqueue_buffers(void) {
   ALint p;
   int s;
-  for (s = 0;  s < ao_data.channels; s++) {
+  for (s = 0;  s < ao_data.channels.num; s++) {
     int till_wrap = NUM_BUF - unqueue_buf[s];
     alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
     if (p >= till_wrap) {
@@ -219,7 +243,7 @@ static void unqueue_buffers(void) {
  * \brief stop playing and empty buffers (for seeking/pause)
  */
 static void reset(void) {
-  alSourceStopv(ao_data.channels, sources);
+  alSourceStopv(ao_data.channels.num, sources);
   unqueue_buffers();
 }
 
@@ -227,14 +251,14 @@ static void reset(void) {
  * \brief stop playing, keep buffers (for pause)
  */
 static void audio_pause(void) {
-  alSourcePausev(ao_data.channels, sources);
+  alSourcePausev(ao_data.channels.num, sources);
 }
 
 /**
  * \brief resume playing, after audio_pause()
  */
 static void audio_resume(void) {
-  alSourcePlayv(ao_data.channels, sources);
+  alSourcePlayv(ao_data.channels.num, sources);
 }
 
 static int get_space(void) {
@@ -243,7 +267,7 @@ static int get_space(void) {
   alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
   queued = NUM_BUF - queued - 3;
   if (queued < 0) return 0;
-  return queued * CHUNK_SIZE * ao_data.channels;
+  return queued * CHUNK_SIZE * ao_data.channels.num;
 }
 
 /**
@@ -254,22 +278,22 @@ static int play(void *data, int len, int flags) {
   int i, j, k;
   int ch;
   int16_t *d = data;
-  len /= ao_data.channels * CHUNK_SIZE;
+  len /= ao_data.channels.num * CHUNK_SIZE;
   for (i = 0; i < len; i++) {
-    for (ch = 0; ch < ao_data.channels; ch++) {
-      for (j = 0, k = ch; j < CHUNK_SIZE / 2; j++, k += ao_data.channels)
+    for (ch = 0; ch < ao_data.channels.num; ch++) {
+      for (j = 0, k = ch; j < CHUNK_SIZE / 2; j++, k += ao_data.channels.num)
         tmpbuf[j] = d[k];
       alBufferData(buffers[ch][cur_buf[ch]], AL_FORMAT_MONO16, tmpbuf,
                      CHUNK_SIZE, ao_data.samplerate);
       alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
       cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
     }
-    d += ao_data.channels * CHUNK_SIZE / 2;
+    d += ao_data.channels.num * CHUNK_SIZE / 2;
   }
   alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
   if (state != AL_PLAYING) // checked here in case of an underrun
-    alSourcePlayv(ao_data.channels, sources);
-  return len * ao_data.channels * CHUNK_SIZE;
+    alSourcePlayv(ao_data.channels.num, sources);
+  return len * ao_data.channels.num * CHUNK_SIZE;
 }
 
 static float get_delay(void) {
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index fa8eccdeea..4ff97b30b6 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -221,12 +221,12 @@ static int control(int cmd,void *arg){
 
 // open & setup audio device
 // return: 1=success 0=fail
-static int init(int rate,int channels,int format,int flags){
+static int init(int rate,const struct mp_chmap *channels,int format,int flags){
   char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
   int oss_format;
   char *mdev = mixer_device, *mchan = mixer_channel;
 
-  mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz  %d chans  %s\n",rate,channels,
+  mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz  %d chans  %s\n",rate,ao_data.channels.num,
     af_fmt2str_short(format));
 
   if (ao_subdevice) {
@@ -339,25 +339,31 @@ ac3_retry:
   mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
     af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
 
-  ao_data.channels = channels;
   if(!AF_FORMAT_IS_AC3(format)) {
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_alsa_def(&sel);
+    if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
+      return 0;
+    int reqchannels = ao_data.channels.num;
     // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
-    if (ao_data.channels > 2) {
-      if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
-	   ao_data.channels != channels ) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", channels);
+    if (reqchannels > 2) {
+      int nchannels = reqchannels;
+      if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1 ||
+	   nchannels != reqchannels ) {
+	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", reqchannels);
 	return 0;
       }
     }
     else {
-      int c = ao_data.channels-1;
+      int c = reqchannels-1;
       if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", ao_data.channels);
+	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", reqchannels);
 	return 0;
       }
-      ao_data.channels=c+1;
+      if (!ao_chmap_sel_get_def(&ao_data, &sel, &ao_data.channels, c + 1))
+        return 0;
     }
-    mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
+    mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels.num, reqchannels);
     // set rate
     ao_data.samplerate=rate;
     ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
@@ -403,7 +409,7 @@ ac3_retry:
 #endif
   }
 
-  ao_data.bps=ao_data.channels;
+  ao_data.bps=ao_data.channels.num;
   switch (ao_data.format & AF_FORMAT_BITS_MASK) {
   case AF_FORMAT_8BIT:
     break;
@@ -459,10 +465,10 @@ static void reset(void){
     ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
   ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
   if(!AF_FORMAT_IS_AC3(ao_data.format)) {
-    if (ao_data.channels > 2)
-      ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
+    if (ao_data.channels.num > 2)
+      ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels.num);
     else {
-      int c = ao_data.channels-1;
+      int c = ao_data.channels.num-1;
       ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
     }
     ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
diff --git a/audio/out/ao_pcm.c b/audio/out/ao_pcm.c
index 5b3cc94395..903e89bc8b 100644
--- a/audio/out/ao_pcm.c
+++ b/audio/out/ao_pcm.c
@@ -69,7 +69,7 @@ static void fput32le(uint32_t val, FILE *fp)
 
 static void write_wave_header(struct ao *ao, FILE *fp, uint64_t data_length)
 {
-    bool use_waveex = ao->channels >= 5 && ao->channels <= 8;
+    bool use_waveex = true;
     uint16_t fmt = ao->format == AF_FORMAT_FLOAT_LE ?
         WAV_ID_FLOAT_PCM : WAV_ID_PCM;
     uint32_t fmt_chunk_size = use_waveex ? 40 : 16;
@@ -86,30 +86,17 @@ static void write_wave_header(struct ao *ao, FILE *fp, uint64_t data_length)
     fput32le(WAV_ID_FMT, fp);
     fput32le(fmt_chunk_size, fp);
     fput16le(use_waveex ? WAV_ID_FORMAT_EXTENSIBLE : fmt, fp);
-    fput16le(ao->channels, fp);
+    fput16le(ao->channels.num, fp);
     fput32le(ao->samplerate, fp);
     fput32le(ao->bps, fp);
-    fput16le(ao->channels * (bits / 8), fp);
+    fput16le(ao->channels.num * (bits / 8), fp);
     fput16le(bits, fp);
 
     if (use_waveex) {
         // Extension chunk
         fput16le(22, fp);
         fput16le(bits, fp);
-        switch (ao->channels) {
-            case 5:
-                fput32le(0x0607, fp); // L R C Lb Rb
-                break;
-            case 6:
-                fput32le(0x060f, fp); // L R C Lb Rb LFE
-                break;
-            case 7:
-                fput32le(0x0727, fp); // L R C Cb Ls Rs LFE
-                break;
-            case 8:
-                fput32le(0x063f, fp); // L R C Lb Rb Ls Rs LFE
-                break;
-        }
+        fput32le(mp_chmap_to_waveext(&ao->channels), fp);
         // 2 bytes format + 14 bytes guid
         fput32le(fmt, fp);
         fput32le(0x00100000, fp);
@@ -159,14 +146,19 @@ static int init(struct ao *ao, char *params)
         }
     }
 
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_waveext(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        return -1;
+
     ao->outburst = 65536;
-    ao->bps = ao->channels * ao->samplerate * (af_fmt2bits(ao->format) / 8);
+    ao->bps = ao->channels.num * ao->samplerate * (af_fmt2bits(ao->format) / 8);
 
     mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\n"
             "PCM: Samplerate: %d Hz   Channels: %d   Format: %s\n",
             priv->outputfilename,
             priv->waveheader ? "WAVE" : "RAW PCM", ao->samplerate,
-            ao->channels, af_fmt2str_short(ao->format));
+            ao->channels.num, af_fmt2str_short(ao->format));
     mp_tmsg(MSGT_AO, MSGL_INFO,
             "[AO PCM] Info: Faster dumping is achieved with -no-video\n"
             "[AO PCM] Info: To write WAVE files use -ao pcm:waveheader (default).\n");
@@ -222,20 +214,12 @@ static int play(struct ao *ao, void *data, int len, int flags)
 {
     struct priv *priv = ao->priv;
 
-    if (ao->channels == 5 || ao->channels == 6 || ao->channels == 8) {
-        int frame_size = af_fmt2bits(ao->format) / 8;
-        len -= len % (frame_size * ao->channels);
-        reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
-                            AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
-                            ao->channels, len / frame_size, frame_size);
-    }
     fwrite(data, len, 1, priv->fp);
     priv->data_length += len;
     return len;
 }
 
 const struct ao_driver audio_out_pcm = {
-    .is_new = true,
     .info = &(const struct ao_info) {
         "RAW PCM/WAVE file writer audio output",
         "pcm",
diff --git a/audio/out/ao_portaudio.c b/audio/out/ao_portaudio.c
index b0744e8f8a..f84447bdb4 100644
--- a/audio/out/ao_portaudio.c
+++ b/audio/out/ao_portaudio.c
@@ -273,9 +273,15 @@ static int init(struct ao *ao, char *params)
     if (pa_device == paNoDevice)
         goto error_exit;
 
+    // The actual channel order probably depends on the platform.
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_waveext_def(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        goto error_exit;
+
     PaStreamParameters sp = {
         .device = pa_device,
-        .channelCount = ao->channels,
+        .channelCount = ao->channels.num,
         .suggestedLatency
             = Pa_GetDeviceInfo(pa_device)->defaultHighOutputLatency,
     };
@@ -298,7 +304,7 @@ static int init(struct ao *ao, char *params)
 
     ao->format = fmt->mp_format;
     sp.sampleFormat = fmt->pa_format;
-    priv->framelen = ao->channels * (af_fmt2bits(ao->format) / 8);
+    priv->framelen = ao->channels.num * (af_fmt2bits(ao->format) / 8);
     ao->bps = ao->samplerate * priv->framelen;
 
     if (!check_pa_ret(Pa_IsFormatSupported(NULL, &sp, ao->samplerate)))
@@ -413,7 +419,6 @@ static void resume(struct ao *ao)
 }
 
 const struct ao_driver audio_out_portaudio = {
-    .is_new = true,
     .info = &(const struct ao_info) {
         "PortAudio",
         "portaudio",
diff --git a/audio/out/ao_pulse.c b/audio/out/ao_pulse.c
index 314ef778bd..15a2b8b0a1 100644
--- a/audio/out/ao_pulse.c
+++ b/audio/out/ao_pulse.c
@@ -143,6 +143,59 @@ static const struct format_map {
     {AF_FORMAT_UNKNOWN, 0}
 };
 
+static const int speaker_map[][2] = {
+  {PA_CHANNEL_POSITION_MONO,                    MP_SPEAKER_ID_FC},
+  {PA_CHANNEL_POSITION_FRONT_LEFT,              MP_SPEAKER_ID_FL},
+  {PA_CHANNEL_POSITION_FRONT_RIGHT,             MP_SPEAKER_ID_FR},
+  {PA_CHANNEL_POSITION_FRONT_CENTER,            MP_SPEAKER_ID_FC},
+  {PA_CHANNEL_POSITION_REAR_CENTER,             MP_SPEAKER_ID_BC},
+  {PA_CHANNEL_POSITION_REAR_LEFT,               MP_SPEAKER_ID_BL},
+  {PA_CHANNEL_POSITION_REAR_RIGHT,              MP_SPEAKER_ID_BR},
+  {PA_CHANNEL_POSITION_LFE,                     MP_SPEAKER_ID_LFE},
+  {PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,    MP_SPEAKER_ID_FLC},
+  {PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,   MP_SPEAKER_ID_FRC},
+  {PA_CHANNEL_POSITION_SIDE_LEFT,               MP_SPEAKER_ID_SL},
+  {PA_CHANNEL_POSITION_SIDE_RIGHT,              MP_SPEAKER_ID_SR},
+  {PA_CHANNEL_POSITION_TOP_CENTER,              MP_SPEAKER_ID_TC},
+  {PA_CHANNEL_POSITION_TOP_FRONT_LEFT,          MP_SPEAKER_ID_TFL},
+  {PA_CHANNEL_POSITION_TOP_FRONT_RIGHT,         MP_SPEAKER_ID_TFR},
+  {PA_CHANNEL_POSITION_TOP_FRONT_CENTER,        MP_SPEAKER_ID_TFC},
+  {PA_CHANNEL_POSITION_TOP_REAR_LEFT,           MP_SPEAKER_ID_TBL},
+  {PA_CHANNEL_POSITION_TOP_REAR_RIGHT,          MP_SPEAKER_ID_TBR},
+  {PA_CHANNEL_POSITION_TOP_REAR_CENTER,         MP_SPEAKER_ID_TBC},
+  {PA_CHANNEL_POSITION_INVALID,                 -1}
+};
+
+static bool chmap_pa_from_mp(pa_channel_map *dst, struct mp_chmap *src)
+{
+    if (src->num > PA_CHANNELS_MAX)
+        return false;
+    dst->channels = src->num;
+    for (int n = 0; n < src->num; n++) {
+        int mp_speaker = src->speaker[n];
+        int pa_speaker = PA_CHANNEL_POSITION_INVALID;
+        for (int i = 0; speaker_map[i][1] != -1; i++) {
+            if (speaker_map[i][1] == mp_speaker) {
+                pa_speaker = speaker_map[i][0];
+                break;
+            }
+        }
+        if (pa_speaker == PA_CHANNEL_POSITION_INVALID)
+            return false;
+        dst->map[n] = pa_speaker;
+    }
+    return true;
+}
+
+static bool select_chmap(struct ao *ao, pa_channel_map *dst)
+{
+    struct mp_chmap_sel sel = {0};
+    for (int n = 0; speaker_map[n][1] != -1; n++)
+        mp_chmap_sel_add_speaker(&sel, speaker_map[n][1]);
+    return ao_chmap_sel_adjust(ao, &sel, &ao->channels) &&
+           chmap_pa_from_mp(dst, &ao->channels);
+}
+
 static void uninit(struct ao *ao, bool cut_audio)
 {
     struct priv *priv = ao->priv;
@@ -209,30 +262,6 @@ static int init(struct ao *ao, char *params)
         priv->broken_pause = true;
     }
 
-    ss.channels = ao->channels;
-    ss.rate = ao->samplerate;
-
-    const struct format_map *fmt_map = format_maps;
-    while (fmt_map->mp_format != ao->format) {
-        if (fmt_map->mp_format == AF_FORMAT_UNKNOWN) {
-            mp_msg(MSGT_AO, MSGL_V,
-                   "AO: [pulse] Unsupported format, using default\n");
-            fmt_map = format_maps;
-            break;
-        }
-        fmt_map++;
-    }
-    ao->format = fmt_map->mp_format;
-    ss.format = fmt_map->pa_format;
-
-    if (!pa_sample_spec_valid(&ss)) {
-        mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Invalid sample spec\n");
-        goto fail;
-    }
-
-    pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_ALSA);
-    ao->bps = pa_bytes_per_second(&ss);
-
     if (!(priv->mainloop = pa_threaded_mainloop_new())) {
         mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Failed to allocate main loop\n");
         goto fail;
@@ -260,6 +289,32 @@ static int init(struct ao *ao, char *params)
     if (pa_context_get_state(priv->context) != PA_CONTEXT_READY)
         goto unlock_and_fail;
 
+    ss.channels = ao->channels.num;
+    ss.rate = ao->samplerate;
+
+    const struct format_map *fmt_map = format_maps;
+    while (fmt_map->mp_format != ao->format) {
+        if (fmt_map->mp_format == AF_FORMAT_UNKNOWN) {
+            mp_msg(MSGT_AO, MSGL_V,
+                   "AO: [pulse] Unsupported format, using default\n");
+            fmt_map = format_maps;
+            break;
+        }
+        fmt_map++;
+    }
+    ao->format = fmt_map->mp_format;
+    ss.format = fmt_map->pa_format;
+
+    if (!pa_sample_spec_valid(&ss)) {
+        mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Invalid sample spec\n");
+        goto fail;
+    }
+
+    if (!select_chmap(ao, &map))
+        goto fail;
+
+    ao->bps = pa_bytes_per_second(&ss);
+
     if (!(priv->stream = pa_stream_new(priv->context, "audio stream", &ss,
                                        &map)))
         goto unlock_and_fail;
@@ -495,7 +550,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
             const ao_control_vol_t *vol = arg;
             struct pa_cvolume volume;
 
-            pa_cvolume_reset(&volume, ao->channels);
+            pa_cvolume_reset(&volume, ao->channels.num);
             if (volume.channels != 2)
                 pa_cvolume_set(&volume, volume.channels, VOL_MP2PA(vol->left));
             else {
@@ -533,7 +588,6 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
 }
 
 const struct ao_driver audio_out_pulse = {
-    .is_new = true,
     .info = &(const struct ao_info) {
         "PulseAudio audio output",
         "pulse",
diff --git a/audio/out/ao_rsound.c b/audio/out/ao_rsound.c
index 7722bf19d2..78ea2c63f5 100644
--- a/audio/out/ao_rsound.c
+++ b/audio/out/ao_rsound.c
@@ -121,8 +121,16 @@ static int init(struct ao *ao, char *params)
         free(port);
     }
 
+    // Actual channel layout unknown.
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_waveext_def(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) {
+        rsd_free(priv->rd);
+        return -1;
+    }
+
     rsd_set_param(priv->rd, RSD_SAMPLERATE, &ao->samplerate);
-    rsd_set_param(priv->rd, RSD_CHANNELS, &ao->channels);
+    rsd_set_param(priv->rd, RSD_CHANNELS, &ao->channels.num);
 
     int rsd_format = set_format(ao);
     rsd_set_param(priv->rd, RSD_FORMAT, &rsd_format);
@@ -132,7 +140,7 @@ static int init(struct ao *ao, char *params)
         return -1;
     }
 
-    ao->bps = ao->channels * ao->samplerate * af_fmt2bits(ao->format) / 8;
+    ao->bps = ao->channels.num * ao->samplerate * af_fmt2bits(ao->format) / 8;
 
     return 0;
 }
@@ -189,7 +197,6 @@ static float get_delay(struct ao *ao)
 }
 
 const struct ao_driver audio_out_rsound = {
-    .is_new    = true,
     .info      = &(const struct ao_info) {
         .name       = "RSound output driver",
         .short_name = "rsound",
diff --git a/audio/out/ao_sdl.c b/audio/out/ao_sdl.c
index 7cfb1ae1e2..6678cd3bd3 100644
--- a/audio/out/ao_sdl.c
+++ b/audio/out/ao_sdl.c
@@ -160,6 +160,13 @@ static int init(struct ao *ao, char *params)
         return -1;
     }
 
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_waveext_def(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) {
+        uninit(ao, true);
+        return -1;
+    }
+
     SDL_AudioSpec desired, obtained;
 
     int bytes = 0;
@@ -185,7 +192,7 @@ static int init(struct ao *ao, char *params)
 #endif
     }
     desired.freq = ao->samplerate;
-    desired.channels = ao->channels;
+    desired.channels = ao->channels.num;
     desired.samples = FFMIN(32768, ceil_power_of_two(ao->samplerate * buflen));
     desired.callback = audio_callback;
     desired.userdata = ao;
@@ -236,9 +243,13 @@ static int init(struct ao *ao, char *params)
             return -1;
     }
 
+    if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, obtained.channels)) {
+        uninit(ao, true);
+        return -1;
+    }
+
     ao->samplerate = obtained.freq;
-    ao->channels = obtained.channels;
-    ao->bps = ao->channels * ao->samplerate * bytes;
+    ao->bps = ao->channels.num * ao->samplerate * bytes;
     ao->buffersize = obtained.size * bufcnt;
     ao->outburst = obtained.size;
     priv->buffer = av_fifo_alloc(ao->buffersize);
@@ -362,7 +373,6 @@ static float get_delay(struct ao *ao)
 }
 
 const struct ao_driver audio_out_sdl = {
-    .is_new = true,
     .info = &(const struct ao_info) {
         "SDL Audio",
         "sdl",
diff --git a/audio/out/audio_out_internal.h b/audio/out/audio_out_internal.h
index f3e92dff66..7b863cfd21 100644
--- a/audio/out/audio_out_internal.h
+++ b/audio/out/audio_out_internal.h
@@ -20,11 +20,12 @@
 #define MPLAYER_AUDIO_OUT_INTERNAL_H
 
 #include "core/options.h"
+#include "ao.h"
 
 // prototypes:
 //static ao_info_t info;
 static int control(int cmd, void *arg);
-static int init(int rate,int channels,int format,int flags);
+static int init(int rate,const struct mp_chmap *channels,int format,int flags);
 static void uninit(int immed);
 static void reset(void);
 static int get_space(void);