Merge branch 'audio_changes'

Conflicts:
	audio/out/ao_lavc.c

62 files changed, 3143 insertions, 3498 deletions

diff --git a/audio/audio.c b/audio/audio.c
new file mode 100644
index 0000000000..c9d5c9231c
--- /dev/null
+++ b/audio/audio.c
@@ -0,0 +1,76 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <assert.h>
+
+#include "core/mp_talloc.h"
+#include "audio.h"
+
+void mp_audio_set_format(struct mp_audio *mpa, int format)
+{
+    mpa->format = format;
+    mpa->bps = af_fmt2bits(format) / 8;
+}
+
+void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels)
+{
+    struct mp_chmap map;
+    mp_chmap_from_channels(&map, num_channels);
+    mp_audio_set_channels(mpa, &map);
+}
+
+// Use old MPlayer/ALSA channel layout.
+void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels)
+{
+    struct mp_chmap map;
+    mp_chmap_from_channels_alsa(&map, num_channels);
+    mp_audio_set_channels(mpa, &map);
+}
+
+void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap)
+{
+    assert(mp_chmap_is_empty(chmap) || mp_chmap_is_valid(chmap));
+    mpa->channels = *chmap;
+    mpa->nch = mpa->channels.num;
+}
+
+void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src)
+{
+    mp_audio_set_format(dst, src->format);
+    mp_audio_set_channels(dst, &src->channels);
+    dst->rate = src->rate;
+}
+
+bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b)
+{
+    return a->format == b->format && a->rate == b->rate &&
+           mp_chmap_equals(&a->channels, &b->channels);
+}
+
+char *mp_audio_fmt_to_str(int srate, const struct mp_chmap *chmap, int format)
+{
+    char *chstr = mp_chmap_to_str(chmap);
+    char *res = talloc_asprintf(NULL, "%dHz %s %dch %s", srate, chstr,
+                                chmap->num, af_fmt2str_short(format));
+    talloc_free(chstr);
+    return res;
+}
+
+char *mp_audio_config_to_str(struct mp_audio *mpa)
+{
+    return mp_audio_fmt_to_str(mpa->rate, &mpa->channels, mpa->format);
+}
diff --git a/audio/audio.h b/audio/audio.h
new file mode 100644
index 0000000000..de35e697c8
--- /dev/null
+++ b/audio/audio.h
@@ -0,0 +1,46 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef MP_AUDIO_H
+#define MP_AUDIO_H
+
+#include "format.h"
+#include "chmap.h"
+
+// Audio data chunk
+struct mp_audio {
+    void *audio; // data buffer
+    int len;    // buffer length (in bytes)
+    int rate;   // sample rate
+    struct mp_chmap channels; // channel layout, use mp_audio_set_*() to set
+    int format; // format (AF_FORMAT_...), use mp_audio_set_format() to set
+    // Redundant fields, for convenience
+    int nch;    // number of channels (redundant with chmap)
+    int bps;    // bytes per sample (redundant with format)
+};
+
+void mp_audio_set_format(struct mp_audio *mpa, int format);
+void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels);
+void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels);
+void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap);
+void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src);
+bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b);
+
+char *mp_audio_fmt_to_str(int srate, const struct mp_chmap *chmap, int format);
+char *mp_audio_config_to_str(struct mp_audio *mpa);
+
+#endif
diff --git a/audio/chmap.c b/audio/chmap.c
new file mode 100644
index 0000000000..fcdb95edb3
--- /dev/null
+++ b/audio/chmap.c
@@ -0,0 +1,486 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdlib.h>
+#include <assert.h>
+
+#include "core/mp_msg.h"
+#include "chmap.h"
+
+// Names taken from libavutil/channel_layout.c (Not accessible by API.)
+// Use of these names is hard-coded in some places (e.g. ao_alsa.c)
+static const char *speaker_names[MP_SPEAKER_ID_COUNT][2] = {
+    [MP_SPEAKER_ID_FL]          = {"fl",   "front left"},
+    [MP_SPEAKER_ID_FR]          = {"fr",   "front right"},
+    [MP_SPEAKER_ID_FC]          = {"fc",   "front center"},
+    [MP_SPEAKER_ID_LFE]         = {"lfe",  "low frequency"},
+    [MP_SPEAKER_ID_BL]          = {"bl",   "back left"},
+    [MP_SPEAKER_ID_BR]          = {"br",   "back right"},
+    [MP_SPEAKER_ID_FLC]         = {"flc",  "front left-of-center"},
+    [MP_SPEAKER_ID_FRC]         = {"frc",  "front right-of-center"},
+    [MP_SPEAKER_ID_BC]          = {"bc",   "back center"},
+    [MP_SPEAKER_ID_SL]          = {"sl",   "side left"},
+    [MP_SPEAKER_ID_SR]          = {"sr",   "side right"},
+    [MP_SPEAKER_ID_TC]          = {"tc",   "top center"},
+    [MP_SPEAKER_ID_TFL]         = {"tfl",  "top front left"},
+    [MP_SPEAKER_ID_TFC]         = {"tfc",  "top front center"},
+    [MP_SPEAKER_ID_TFR]         = {"tfr",  "top front right"},
+    [MP_SPEAKER_ID_TBL]         = {"tbl",  "top back left"},
+    [MP_SPEAKER_ID_TBC]         = {"tbc",  "top back center"},
+    [MP_SPEAKER_ID_TBR]         = {"tbr",  "top back right"},
+    [MP_SPEAKER_ID_DL]          = {"dl",   "downmix left"},
+    [MP_SPEAKER_ID_DR]          = {"dr",   "downmix right"},
+    [MP_SPEAKER_ID_WL]          = {"wl",   "wide left"},
+    [MP_SPEAKER_ID_WR]          = {"wr",   "wide right"},
+    [MP_SPEAKER_ID_SDL]         = {"sdl",  "surround direct left"},
+    [MP_SPEAKER_ID_SDR]         = {"sdr",  "surround direct right"},
+    [MP_SPEAKER_ID_LFE2]        = {"lfe2", "low frequency 2"},
+};
+
+// Names taken from libavutil/channel_layout.c (Not accessible by API.)
+// Channel order corresponds to lavc/waveex, except for the alsa entries.
+static const char *std_layout_names[][2] = {
+    {"empty",           ""}, // not in lavc
+    {"mono",            "fc"},
+    {"stereo",          "fl-fr"},
+    {"2.1",             "fl-fr-lfe"},
+    {"3.0",             "fl-fr-fc"},
+    {"3.0(back)",       "fl-fr-bc"},
+    {"4.0",             "fl-fr-fc-bc"},
+    {"quad",            "fl-fr-bl-br"},
+    {"quad(side)",      "fl-fr-sl-sr"},
+    {"3.1",             "fl-fr-fc-lfe"},
+    {"5.0",             "fl-fr-fc-bl-br"},
+    {"5.0(alsa)",       "fl-fr-bl-br-fc"}, // not in lavc
+    {"5.0(side)",       "fl-fr-fc-sl-sr"},
+    {"4.1",             "fl-fr-fc-lfe-bc"},
+    {"4.1(alsa)",       "fl-fr-bl-br-lfe"}, // not in lavc
+    {"5.1",             "fl-fr-fc-lfe-bl-br"},
+    {"5.1(alsa)",       "fl-fr-bl-br-fc-lfe"}, // not in lavc
+    {"5.1(side)",       "fl-fr-fc-lfe-sl-sr"},
+    {"6.0",             "fl-fr-fc-bc-sl-sr"},
+    {"6.0(front)",      "fl-fr-flc-frc-sl-sr"},
+    {"hexagonal",       "fl-fr-fc-bl-br-bc"},
+    {"6.1",             "fl-fr-fc-lfe-bl-br-bc"},
+    {"6.1(front)",      "fl-fr-lfe-flc-frc-sl-sr"},
+    {"7.0",             "fl-fr-fc-bl-br-sl-sr"},
+    {"7.0(front)",      "fl-fr-fc-flc-frc-sl-sr"},
+    {"7.1",             "fl-fr-fc-lfe-bl-br-sl-sr"},
+    {"7.1(alsa)",       "fl-fr-bl-br-fc-lfe-sl-sr"}, // not in lavc
+    {"7.1(wide)",       "fl-fr-fc-lfe-bl-br-flc-frc"},
+    {"7.1(wide-side)",  "fl-fr-fc-lfe-flc-frc-sl-sr"},
+    {"octagonal",       "fl-fr-fc-bl-br-bc-sl-sr"},
+    {"downmix",         "dl-dr"},
+    {0}
+};
+
+static const struct mp_chmap default_layouts[MP_NUM_CHANNELS + 1] = {
+    {0},                                        // empty
+    MP_CHMAP_INIT_MONO,                         // mono
+    MP_CHMAP2(FL, FR),                          // stereo
+    MP_CHMAP3(FL, FR, LFE),                     // 2.1
+    MP_CHMAP4(FL, FR, FC, BC),                  // 4.0
+    MP_CHMAP5(FL, FR, FC, BL,  BR),             // 5.0
+    MP_CHMAP6(FL, FR, FC, LFE, BL, BR),         // 5.1
+    MP_CHMAP7(FL, FR, FC, LFE, BL, BR, BC),     // 6.1
+    MP_CHMAP8(FL, FR, FC, LFE, BL, BR, SL, SR), // 7.1
+};
+
+// The channel order was lavc/waveex, but differs from lavc for 5, 6 and 8
+// channels. 3 and 7 channels were likely undefined (no ALSA support).
+static const char *mplayer_layouts[MP_NUM_CHANNELS + 1] = {
+    [1] = "mono",
+    [2] = "stereo",
+    [4] = "4.0",
+    [5] = "5.0(alsa)",
+    [6] = "5.1(alsa)",
+    [8] = "7.1(alsa)",
+};
+
+// Returns true if speakers are mapped uniquely, and there's at least 1 channel.
+bool mp_chmap_is_valid(const struct mp_chmap *src)
+{
+    bool mapped[MP_SPEAKER_ID_COUNT] = {0};
+    for (int n = 0; n < src->num; n++) {
+        int sp = src->speaker[n];
+        if (sp >= MP_SPEAKER_ID_COUNT || mapped[sp])
+            return false;
+        mapped[sp] = true;
+    }
+    return src->num > 0;
+}
+
+bool mp_chmap_is_empty(const struct mp_chmap *src)
+{
+    return src->num == 0;
+}
+
+// Return true if the channel map defines the number of the channels only, and
+// the channels have to meaning associated with them.
+bool mp_chmap_is_unknown(const struct mp_chmap *src)
+{
+    for (int n = 0; n < src->num; n++) {
+        int speaker = src->speaker[n];
+        if (speaker >= MP_SPEAKER_ID_UNKNOWN0 &&
+            speaker <= MP_SPEAKER_ID_UNKNOWN_LAST)
+            return true;
+    }
+    return false;
+}
+
+// Note: empty channel maps compare as equal. Invalid ones can equal too.
+bool mp_chmap_equals(const struct mp_chmap *a, const struct mp_chmap *b)
+{
+    if (a->num != b->num)
+        return false;
+    for (int n = 0; n < a->num; n++) {
+        if (a->speaker[n] != b->speaker[n])
+            return false;
+    }
+    return true;
+}
+
+// Whether they use the same speakers (even if in different order).
+bool mp_chmap_equals_reordered(const struct mp_chmap *a, const struct mp_chmap *b)
+{
+    struct mp_chmap t1 = *a, t2 = *b;
+    mp_chmap_reorder_norm(&t1);
+    mp_chmap_reorder_norm(&t2);
+    return mp_chmap_equals(&t1, &t2);
+}
+
+bool mp_chmap_is_compatible(const struct mp_chmap *a, const struct mp_chmap *b)
+{
+    if (mp_chmap_equals(a, b))
+        return true;
+    if (a->num == b->num && (mp_chmap_is_unknown(a) || mp_chmap_is_unknown(b)))
+        return true;
+    return false;
+}
+
+bool mp_chmap_is_stereo(const struct mp_chmap *src)
+{
+    static const struct mp_chmap stereo = MP_CHMAP_INIT_STEREO;
+    return mp_chmap_equals(src, &stereo);
+}
+
+static int comp_uint8(const void *a, const void *b)
+{
+    return *(const uint8_t *)a - *(const uint8_t *)b;
+}
+
+// Reorder channels to normal order, with monotonically increasing speaker IDs.
+// We define this order as the same order used with waveex.
+void mp_chmap_reorder_norm(struct mp_chmap *map)
+{
+    uint8_t *arr = &map->speaker[0];
+    qsort(arr, map->num, 1, comp_uint8);
+}
+
+// Set *dst to a standard layout with the given number of channels.
+// If the number of channels is invalid, an invalid map is set, and
+// mp_chmap_is_valid(dst) will return false.
+void mp_chmap_from_channels(struct mp_chmap *dst, int num_channels)
+{
+    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
+        *dst = (struct mp_chmap) {0};
+    } else {
+        *dst = default_layouts[num_channels];
+    }
+}
+
+// Try to do what mplayer/mplayer2/mpv did before channel layouts were
+// introduced, i.e. get the old default channel order.
+void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels)
+{
+    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
+        *dst = (struct mp_chmap) {0};
+    } else {
+        mp_chmap_from_str(dst, bstr0(mplayer_layouts[num_channels]));
+        if (!dst->num)
+            mp_chmap_from_channels(dst, num_channels);
+    }
+}
+
+// Set *dst to an unknown layout for the given numbers of channels.
+// If the number of channels is invalid, an invalid map is set, and
+// mp_chmap_is_valid(dst) will return false.
+void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels)
+{
+    if (num_channels < 0 || num_channels > MP_NUM_CHANNELS) {
+        *dst = (struct mp_chmap) {0};
+    } else {
+        dst->num = num_channels;
+        for (int n = 0; n < dst->num; n++)
+            dst->speaker[n] = MP_SPEAKER_ID_UNKNOWN0 + n;
+    }
+}
+
+// Return channel index of the given speaker, or -1.
+static int mp_chmap_find_speaker(const struct mp_chmap *map, int speaker)
+{
+    for (int n = 0; n < map->num; n++) {
+        if (map->speaker[n] == speaker)
+            return n;
+    }
+    return -1;
+}
+
+static void mp_chmap_remove_speaker(struct mp_chmap *map, int speaker)
+{
+    int index = mp_chmap_find_speaker(map, speaker);
+    if (index >= 0) {
+        for (int n = index; n < map->num - 1; n++)
+            map->speaker[n] = map->speaker[n + 1];
+        map->num--;
+    }
+}
+
+// Some decoders output additional, redundant channels, which are usually
+// useless and will mess up proper audio output channel handling.
+// map: channel map from which the channels should be removed
+// requested: if not NULL, and if it contains any of the "useless" channels,
+//            don't remove them (this is for convenience)
+void mp_chmap_remove_useless_channels(struct mp_chmap *map,
+                                      const struct mp_chmap *requested)
+{
+    if (requested &&
+        mp_chmap_find_speaker(requested, MP_SPEAKER_ID_DL) >= 0)
+        return;
+
+    if (map->num > 2) {
+        mp_chmap_remove_speaker(map, MP_SPEAKER_ID_DL);
+        mp_chmap_remove_speaker(map, MP_SPEAKER_ID_DR);
+    }
+}
+
+// Return the ffmpeg/libav channel layout as in <libavutil/channel_layout.h>.
+// Warning: this ignores the order of the channels, and will return a channel
+//          mask even if the order is different from libavcodec's.
+uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src)
+{
+    // lavc has no concept for unknown layouts yet, so pick a default
+    struct mp_chmap t = *src;
+    if (mp_chmap_is_unknown(&t))
+        mp_chmap_from_channels(&t, t.num);
+    uint64_t mask = 0;
+    for (int n = 0; n < t.num; n++)
+        mask |= 1ULL << t.speaker[n];
+    return mask;
+}
+
+// Return the ffmpeg/libav channel layout as in <libavutil/channel_layout.h>.
+// Returns 0 if the channel order doesn't match lavc's or if it's invalid.
+uint64_t mp_chmap_to_lavc(const struct mp_chmap *src)
+{
+    if (!mp_chmap_is_lavc(src))
+        return 0;
+    return mp_chmap_to_lavc_unchecked(src);
+}
+
+// Set channel map from the ffmpeg/libav channel layout as in
+// <libavutil/channel_layout.h>.
+// If the number of channels exceed MP_NUM_CHANNELS, set dst to empty.
+void mp_chmap_from_lavc(struct mp_chmap *dst, uint64_t src)
+{
+    dst->num = 0;
+    for (int n = 0; n < 64; n++) {
+        if (src & (1ULL << n)) {
+            if (dst->num >= MP_NUM_CHANNELS) {
+                dst->num = 0;
+                return;
+            }
+            dst->speaker[dst->num] = n;
+            dst->num++;
+        }
+    }
+}
+
+bool mp_chmap_is_lavc(const struct mp_chmap *src)
+{
+    if (!mp_chmap_is_valid(src))
+        return false;
+    if (mp_chmap_is_unknown(src))
+        return true;
+    // lavc's channel layout is a bit mask, and channels are always ordered
+    // from LSB to MSB speaker bits, so speaker IDs have to increase.
+    assert(src->num > 0);
+    for (int n = 1; n < src->num; n++) {
+        if (src->speaker[n - 1] >= src->speaker[n])
+            return false;
+    }
+    for (int n = 0; n < src->num; n++) {
+        if (src->speaker[n] >= 64)
+            return false;
+    }
+    return true;
+}
+
+void mp_chmap_reorder_to_lavc(struct mp_chmap *map)
+{
+    if (!mp_chmap_is_valid(map))
+        return;
+    uint64_t mask = mp_chmap_to_lavc_unchecked(map);
+    mp_chmap_from_lavc(map, mask);
+}
+
+// Get reordering array for from->to reordering. from->to must have the same set
+// of speakers (i.e. same number and speaker IDs, just different order). Then,
+// for each speaker n, dst[n] will be set such that:
+//      to->speaker[dst[n]] = from->speaker[n]
+// (dst[n] gives the source channel for destination channel n)
+void mp_chmap_get_reorder(int dst[MP_NUM_CHANNELS], const struct mp_chmap *from,
+                          const struct mp_chmap *to)
+{
+    assert(from->num == to->num);
+    if (mp_chmap_is_unknown(from) || mp_chmap_is_unknown(to)) {
+        for (int n = 0; n < from->num; n++)
+            dst[n] = n;
+        return;
+    }
+    // Same set of speakers required
+    assert(mp_chmap_equals_reordered(from, to));
+    for (int n = 0; n < from->num; n++) {
+        int src = from->speaker[n];
+        dst[n] = -1;
+        for (int i = 0; i < to->num; i++) {
+            if (src == to->speaker[i]) {
+                dst[n] = i;
+                break;
+            }
+        }
+        assert(dst[n] != -1);
+    }
+    for (int n = 0; n < from->num; n++)
+        assert(to->speaker[dst[n]] == from->speaker[n]);
+}
+
+// Returns something like "fl-fr-fc". If there's a standard layout in lavc
+// order, return that, e.g. "3.0" instead of "fl-fr-fc".
+// Unassigned but valid speakers get names like "sp28".
+char *mp_chmap_to_str(const struct mp_chmap *src)
+{
+    char *res = talloc_strdup(NULL, "");
+
+    if (mp_chmap_is_unknown(src))
+        return talloc_asprintf_append_buffer(res, "unknown%d", src->num);
+
+    for (int n = 0; n < src->num; n++) {
+        int sp = src->speaker[n];
+        const char *s = sp < MP_SPEAKER_ID_COUNT ? speaker_names[sp][0] : NULL;
+        char buf[10];
+        if (!s) {
+            snprintf(buf, sizeof(buf), "sp%d", sp);
+            s = buf;
+        }
+        res = talloc_asprintf_append_buffer(res, "%s%s", n > 0 ? "-" : "", s);
+    }
+
+    // To standard layout name
+    for (int n = 0; std_layout_names[n][0]; n++) {
+        if (res && strcmp(res, std_layout_names[n][1]) == 0) {
+            talloc_free(res);
+            res = talloc_strdup(NULL, std_layout_names[n][0]);
+            break;
+        }
+    }
+
+    return res;
+}
+
+// If src can be parsed as channel map (as produced by mp_chmap_to_str()),
+// return true and set *dst. Otherwise, return false and don't change *dst.
+// Note: call mp_chmap_is_valid() to test whether the returned map is valid
+//       the map could be empty, or contain multiply mapped channels
+bool mp_chmap_from_str(struct mp_chmap *dst, bstr src)
+{
+    // Single number corresponds to mp_chmap_from_channels()
+    if (src.len > 0) {
+        bstr t = src;
+        bool unknown = bstr_eatstart0(&t, "unknown");
+        bstr rest;
+        long long count = bstrtoll(t, &rest, 10);
+        if (rest.len == 0) {
+            struct mp_chmap res;
+            if (unknown) {
+                mp_chmap_set_unknown(&res, count);
+            } else {
+                mp_chmap_from_channels(&res, count);
+            }
+            if (mp_chmap_is_valid(&res)) {
+                *dst = res;
+                return true;
+            }
+        }
+    }
+
+    // From standard layout name
+    for (int n = 0; std_layout_names[n][0]; n++) {
+        if (bstr_equals0(src, std_layout_names[n][0])) {
+            src = bstr0(std_layout_names[n][1]);
+            break;
+        }
+    }
+
+    // Explicit speaker list (separated by "-")
+    struct mp_chmap res = {0};
+    while (src.len) {
+        bstr s;
+        bstr_split_tok(src, "-", &s, &src);
+        int speaker = -1;
+        for (int n = 0; n < MP_SPEAKER_ID_COUNT; n++) {
+            const char *name = speaker_names[n][0];
+            if (name && bstr_equals0(s, name)) {
+                speaker = n;
+                break;
+            }
+        }
+        if (speaker < 0) {
+            if (bstr_eatstart0(&s, "sp")) {
+                long long sp = bstrtoll(s, &s, 0);
+                if (s.len == 0 && sp >= 0 && sp < MP_SPEAKER_ID_COUNT)
+                    speaker = sp;
+            }
+            if (speaker < 0)
+                return false;
+        }
+        if (res.num >= MP_NUM_CHANNELS)
+            return false;
+        res.speaker[res.num] = speaker;
+        res.num++;
+    }
+
+    *dst = res;
+    return true;
+}
+
+void mp_chmap_print_help(int msgt, int msgl)
+{
+    mp_msg(msgt, msgl, "Speakers:\n");
+    for (int n = 0; n < MP_SPEAKER_ID_COUNT; n++) {
+        if (speaker_names[n][0])
+            mp_msg(msgt, msgl, "    %-16s (%s)\n",
+                   speaker_names[n][0], speaker_names[n][1]);
+    }
+    mp_msg(msgt, msgl, "Standard layouts:\n");
+    for (int n = 0; std_layout_names[n][0]; n++) {
+        mp_msg(msgt, msgl, "    %-16s (%s)\n",
+               std_layout_names[n][0], std_layout_names[n][1]);
+    }
+    for (int n = 0; n < MP_NUM_CHANNELS; n++)
+        mp_msg(msgt, msgl, "    unknown%d\n", n);
+}
diff --git a/audio/chmap.h b/audio/chmap.h
new file mode 100644
index 0000000000..1848c86efd
--- /dev/null
+++ b/audio/chmap.h
@@ -0,0 +1,132 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef MP_CHMAP_H
+#define MP_CHMAP_H
+
+#include <inttypes.h>
+#include <stdbool.h>
+#include "core/bstr.h"
+
+#define MP_NUM_CHANNELS 8
+
+// Speaker a channel can be assigned to.
+// This corresponds to WAVEFORMATEXTENSIBLE channel mask bit indexes.
+// E.g. channel_mask = (1 << MP_SPEAKER_ID_FL) | ...
+enum {
+    // Official WAVEFORMATEXTENSIBLE (shortened names)
+    MP_SPEAKER_ID_FL = 0,       // FRONT_LEFT
+    MP_SPEAKER_ID_FR,           // FRONT_RIGHT
+    MP_SPEAKER_ID_FC,           // FRONT_CENTER
+    MP_SPEAKER_ID_LFE,          // LOW_FREQUENCY
+    MP_SPEAKER_ID_BL,           // BACK_LEFT
+    MP_SPEAKER_ID_BR,           // BACK_RIGHT
+    MP_SPEAKER_ID_FLC,          // FRONT_LEFT_OF_CENTER
+    MP_SPEAKER_ID_FRC,          // FRONT_RIGHT_OF_CENTER
+    MP_SPEAKER_ID_BC,           // BACK_CENTER
+    MP_SPEAKER_ID_SL,           // SIDE_LEFT
+    MP_SPEAKER_ID_SR,           // SIDE_RIGHT
+    MP_SPEAKER_ID_TC,           // TOP_CENTER
+    MP_SPEAKER_ID_TFL,          // TOP_FRONT_LEFT
+    MP_SPEAKER_ID_TFC,          // TOP_FRONT_CENTER
+    MP_SPEAKER_ID_TFR,          // TOP_FRONT_RIGHT
+    MP_SPEAKER_ID_TBL,          // TOP_BACK_LEFT
+    MP_SPEAKER_ID_TBC,          // TOP_BACK_CENTER
+    MP_SPEAKER_ID_TBR,          // TOP_BACK_RIGHT
+     // Inofficial/libav* extensions
+    MP_SPEAKER_ID_DL = 29,      // STEREO_LEFT (stereo downmix special speakers)
+    MP_SPEAKER_ID_DR,           // STEREO_RIGHT
+    MP_SPEAKER_ID_WL,           // WIDE_LEFT
+    MP_SPEAKER_ID_WR,           // WIDE_RIGHT
+    MP_SPEAKER_ID_SDL,          // SURROUND_DIRECT_LEFT
+    MP_SPEAKER_ID_SDR,          // SURROUND_DIRECT_RIGHT
+    MP_SPEAKER_ID_LFE2,         // LOW_FREQUENCY_2
+
+    // Special mpv-specific speaker entries reserved for channels which have no
+    // known meaning.
+    MP_SPEAKER_ID_UNKNOWN0 = 64,
+    MP_SPEAKER_ID_UNKNOWN_LAST = MP_SPEAKER_ID_UNKNOWN0 + MP_NUM_CHANNELS - 1,
+
+    // Including the unassigned IDs in between. This is not a valid ID anymore.
+    MP_SPEAKER_ID_COUNT,
+};
+
+struct mp_chmap {
+    uint8_t num; // number of channels
+    // Given a channel n, speaker[n] is the speaker ID driven by that channel.
+    // Entries after speaker[num - 1] are undefined.
+    uint8_t speaker[MP_NUM_CHANNELS];
+};
+
+#define MP_SP(speaker) MP_SPEAKER_ID_ ## speaker
+
+#define MP_CHMAP2(a, b) \
+    {2, {MP_SP(a), MP_SP(b)}}
+#define MP_CHMAP3(a, b, c) \
+    {3, {MP_SP(a), MP_SP(b), MP_SP(c)}}
+#define MP_CHMAP4(a, b, c, d) \
+    {4, {MP_SP(a), MP_SP(b), MP_SP(c), MP_SP(d)}}
+#define MP_CHMAP5(a, b, c, d, e) \
+    {5, {MP_SP(a), MP_SP(b), MP_SP(c), MP_SP(d), MP_SP(e)}}
+#define MP_CHMAP6(a, b, c, d, e, f) \
+    {6, {MP_SP(a), MP_SP(b), MP_SP(c), MP_SP(d), MP_SP(e), MP_SP(f)}}
+#define MP_CHMAP7(a, b, c, d, e, f, g) \
+    {7, {MP_SP(a), MP_SP(b), MP_SP(c), MP_SP(d), MP_SP(e), MP_SP(f), MP_SP(g)}}
+#define MP_CHMAP8(a, b, c, d, e, f, g, h) \
+    {8, {MP_SP(a), MP_SP(b), MP_SP(c), MP_SP(d), MP_SP(e), MP_SP(f), MP_SP(g), MP_SP(h)}}
+
+#define MP_CHMAP_INIT_MONO {1, {MP_SPEAKER_ID_FC}}
+#define MP_CHMAP_INIT_STEREO MP_CHMAP2(FL, FR)
+
+bool mp_chmap_is_valid(const struct mp_chmap *src);
+bool mp_chmap_is_empty(const struct mp_chmap *src);
+bool mp_chmap_is_unknown(const struct mp_chmap *src);
+bool mp_chmap_equals(const struct mp_chmap *a, const struct mp_chmap *b);
+bool mp_chmap_equals_reordered(const struct mp_chmap *a, const struct mp_chmap *b);
+bool mp_chmap_is_compatible(const struct mp_chmap *a, const struct mp_chmap *b);
+bool mp_chmap_is_stereo(const struct mp_chmap *src);
+
+void mp_chmap_reorder_norm(struct mp_chmap *map);
+
+void mp_chmap_from_channels(struct mp_chmap *dst, int num_channels);
+void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels);
+void mp_chmap_from_channels_alsa(struct mp_chmap *dst, int num_channels);
+
+void mp_chmap_remove_useless_channels(struct mp_chmap *map,
+                                      const struct mp_chmap *requested);
+
+uint64_t mp_chmap_to_lavc(const struct mp_chmap *src);
+uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src);
+void mp_chmap_from_lavc(struct mp_chmap *dst, uint64_t src);
+
+bool mp_chmap_is_lavc(const struct mp_chmap *src);
+void mp_chmap_reorder_to_lavc(struct mp_chmap *map);
+
+void mp_chmap_get_reorder(int dst[MP_NUM_CHANNELS], const struct mp_chmap *from,
+                          const struct mp_chmap *to);
+
+char *mp_chmap_to_str(const struct mp_chmap *src);
+bool mp_chmap_from_str(struct mp_chmap *dst, bstr src);
+void mp_chmap_print_help(int msgt, int msgl);
+
+// Use these to avoid chaos in case lavc's definition should diverge from MS.
+#define mp_chmap_to_waveext mp_chmap_to_lavc
+#define mp_chmap_from_waveext mp_chmap_from_lavc
+#define mp_chmap_is_waveext mp_chmap_is_lavc
+#define mp_chmap_reorder_to_waveext mp_chmap_reorder_to_lavc
+
+#endif
diff --git a/audio/chmap_sel.c b/audio/chmap_sel.c
new file mode 100644
index 0000000000..8e5be5c86e
--- /dev/null
+++ b/audio/chmap_sel.c
@@ -0,0 +1,210 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdlib.h>
+#include <assert.h>
+
+#include "chmap_sel.h"
+
+// 5.1 and 5.1(side) are practically the same. It doesn't make much sense to
+// reject either of them.
+static const int replaceable_speakers[][2] = {
+    {MP_SPEAKER_ID_SL, MP_SPEAKER_ID_BL},
+    {MP_SPEAKER_ID_SR, MP_SPEAKER_ID_BR},
+    {-1},
+};
+
+// list[] contains a list of speaker pairs, with each pair indicating how
+// a speaker can be swapped for another speaker. Try to replace speakers from
+// the left of the list with the ones on the right, or the other way around.
+static bool replace_speakers(struct mp_chmap *map, const int list[][2])
+{
+    if (!mp_chmap_is_valid(map))
+        return false;
+    for (int dir = 0; dir < 2; dir++) {
+        int from = dir ? 0 : 1;
+        int to   = dir ? 1 : 0;
+        bool replaced = false;
+        struct mp_chmap t = *map;
+        for (int n = 0; n < t.num; n++) {
+            for (int i = 0; list[i][0] != -1; i++) {
+                if (t.speaker[n] == list[i][from]) {
+                    t.speaker[n] = list[i][to];
+                    replaced = true;
+                    break;
+                }
+            }
+        }
+        if (replaced && mp_chmap_is_valid(&t)) {
+            *map = t;
+            return true;
+        }
+    }
+    return false;
+}
+
+// Allow all channel layouts that can be expressed with mp_chmap.
+// (By default, all layouts are rejected.)
+void mp_chmap_sel_add_any(struct mp_chmap_sel *s)
+{
+    s->allow_any = true;
+}
+
+// Allow all waveext formats, and force waveext channel order.
+void mp_chmap_sel_add_waveext(struct mp_chmap_sel *s)
+{
+    s->allow_waveext = true;
+}
+
+void mp_chmap_sel_add_alsa_def(struct mp_chmap_sel *s)
+{
+    for (int n = 0; n < MP_NUM_CHANNELS; n++) {
+        struct mp_chmap t;
+        mp_chmap_from_channels_alsa(&t, n);
+        if (t.num)
+            mp_chmap_sel_add_map(s, &t);
+    }
+}
+
+#define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0]))
+
+// Add a channel map that should be allowed.
+void mp_chmap_sel_add_map(struct mp_chmap_sel *s, const struct mp_chmap *map)
+{
+    assert(s->num_chmaps < ARRAY_LEN(s->chmaps));
+    if (mp_chmap_is_valid(map))
+        s->chmaps[s->num_chmaps++] = *map;
+}
+
+// Allow all waveext formats in default order.
+void mp_chmap_sel_add_waveext_def(struct mp_chmap_sel *s)
+{
+    for (int n = 1; n < MP_NUM_CHANNELS; n++) {
+        struct mp_chmap map;
+        mp_chmap_from_channels(&map, n);
+        mp_chmap_sel_add_map(s, &map);
+    }
+}
+
+// Whitelist a speaker (MP_SPEAKER_ID_...). All layouts that contain whitelisted
+// speakers are allowed.
+void mp_chmap_sel_add_speaker(struct mp_chmap_sel *s, int id)
+{
+    assert(id >= 0 && id < MP_SPEAKER_ID_COUNT);
+    s->speakers[id] = true;
+}
+
+static bool test_speakers(const struct mp_chmap_sel *s, struct mp_chmap *map)
+{
+    for (int n = 0; n < map->num; n++) {
+        if (!s->speakers[map->speaker[n]])
+            return false;
+    }
+    return true;
+}
+
+static bool test_maps(const struct mp_chmap_sel *s, struct mp_chmap *map)
+{
+    for (int n = 0; n < s->num_chmaps; n++) {
+        if (mp_chmap_equals_reordered(&s->chmaps[n], map)) {
+            *map = s->chmaps[n];
+            return true;
+        }
+    }
+    return false;
+}
+
+static bool test_waveext(const struct mp_chmap_sel *s, struct mp_chmap *map)
+{
+    if (s->allow_waveext) {
+        struct mp_chmap t = *map;
+        mp_chmap_reorder_to_waveext(&t);
+        if (mp_chmap_is_waveext(&t)) {
+            *map = t;
+            return true;
+        }
+    }
+    return false;
+}
+
+static bool test_layout(const struct mp_chmap_sel *s, struct mp_chmap *map)
+{
+    if (!mp_chmap_is_valid(map))
+        return false;
+
+    return s->allow_any || test_waveext(s, map) || test_speakers(s, map) ||
+           test_maps(s, map);
+}
+
+// Determine which channel map to use given a source channel map, and various
+// parameters restricting possible choices. If the map doesn't match, select
+// a fallback and set it.
+// If no matching layout is found, a reordered layout may be returned.
+// If that is not possible, a fallback for up/downmixing may be returned.
+// If no choice is possible, set *map to empty.
+bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map)
+{
+    if (test_layout(s, map))
+        return true;
+    if (mp_chmap_is_unknown(map)) {
+        struct mp_chmap t = {0};
+        if (mp_chmap_sel_get_def(s, &t, map->num) && test_layout(s, &t)) {
+            *map = t;
+            return true;
+        }
+    }
+    // 5.1 <-> 5.1(side)
+    if (replace_speakers(map, replaceable_speakers) && test_layout(s, map))
+        return true;
+    // Fallback to mono/stereo as last resort
+    if (map->num == 1) {
+        *map = (struct mp_chmap) MP_CHMAP_INIT_MONO;
+    } else if (map->num >= 2) {
+        *map = (struct mp_chmap) MP_CHMAP_INIT_STEREO;
+    }
+    if (test_layout(s, map))
+        return true;
+    *map = (struct mp_chmap) {0};
+    return false;
+}
+
+// Set map to a default layout with num channels. Used for audio APIs that
+// return a channel count as part of format negotiation, but give no
+// information about the channel layout.
+// If the channel count is correct, do nothing and leave *map untouched.
+bool mp_chmap_sel_get_def(const struct mp_chmap_sel *s, struct mp_chmap *map,
+                          int num)
+{
+    if (map->num != num) {
+        *map = (struct mp_chmap) {0};
+        // Set of speakers or waveext might allow it.
+        struct mp_chmap t;
+        mp_chmap_from_channels(&t, num);
+        mp_chmap_reorder_to_waveext(&t);
+        if (test_layout(s, &t)) {
+            *map = t;
+        } else {
+            for (int n = 0; n < s->num_chmaps; n++) {
+                if (s->chmaps[n].num == num) {
+                    *map = s->chmaps[n];
+                    break;
+                }
+            }
+        }
+    }
+    return map->num > 0;
+}
diff --git a/audio/chmap_sel.h b/audio/chmap_sel.h
new file mode 100644
index 0000000000..c9d75196a5
--- /dev/null
+++ b/audio/chmap_sel.h
@@ -0,0 +1,43 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef MP_CHMAP_SEL_H
+#define MP_CHMAP_SEL_H
+
+#include <stdbool.h>
+
+#include "chmap.h"
+
+struct mp_chmap_sel {
+    // should be considered opaque
+    bool allow_any, allow_waveext;
+    bool speakers[MP_SPEAKER_ID_COUNT];
+    struct mp_chmap chmaps[20];
+    int num_chmaps;
+};
+
+void mp_chmap_sel_add_any(struct mp_chmap_sel *s);
+void mp_chmap_sel_add_waveext(struct mp_chmap_sel *s);
+void mp_chmap_sel_add_waveext_def(struct mp_chmap_sel *s);
+void mp_chmap_sel_add_alsa_def(struct mp_chmap_sel *s);
+void mp_chmap_sel_add_map(struct mp_chmap_sel *s, const struct mp_chmap *map);
+void mp_chmap_sel_add_speaker(struct mp_chmap_sel *s, int id);
+bool mp_chmap_sel_adjust(const struct mp_chmap_sel *s, struct mp_chmap *map);
+bool mp_chmap_sel_get_def(const struct mp_chmap_sel *s, struct mp_chmap *map,
+                          int num);
+
+#endif
diff --git a/audio/decode/ad.h b/audio/decode/ad.h
index 0bbc11ed9b..ff51ecbe35 100644
--- a/audio/decode/ad.h
+++ b/audio/decode/ad.h
@@ -48,10 +48,4 @@ extern const ad_functions_t * const mpcodecs_ad_drivers[];
 // fallback if ADCTRL_SKIP not implemented: ds_fill_buffer(sh_audio->ds);
 #define ADCTRL_SKIP_FRAME 2      // skip block/frame, called while seeking
 
-// fallback if ADCTRL_QUERY_FORMAT not implemented: sh_audio->sample_format
-#define ADCTRL_QUERY_FORMAT 3    // test for availabilty of a format
-
-// fallback: use hw mixer in libao
-#define ADCTRL_SET_VOLUME 4      // not used at the moment
-
 #endif /* MPLAYER_AD_H */
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index 248df307d7..8abc0a6035 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -32,9 +32,11 @@
 #include "core/codecs.h"
 #include "core/mp_msg.h"
 #include "core/options.h"
+#include "core/av_opts.h"
 
 #include "ad_internal.h"
 #include "audio/reorder_ch.h"
+#include "audio/fmt-conversion.h"
 
 #include "compat/mpbswap.h"
 #include "compat/libav.h"
@@ -49,6 +51,16 @@ struct priv {
     int output_left;
     int unitsize;
     int previous_data_left;  // input demuxer packet data
+    bool force_channel_map;
+};
+
+#define OPT_BASE_STRUCT struct MPOpts
+
+const m_option_t ad_lavc_decode_opts_conf[] = {
+    OPT_FLOATRANGE("ac3drc", ad_lavc_param.ac3drc, 0, 0, 2),
+    OPT_FLAG("downmix", ad_lavc_param.downmix, 0),
+    OPT_STRING("o", ad_lavc_param.avopt, 0),
+    {0}
 };
 
 struct pcm_map
@@ -144,17 +156,9 @@ static int preinit(sh_audio_t *sh)
 static int setup_format(sh_audio_t *sh_audio,
                         const AVCodecContext *lavc_context)
 {
-    int sample_format = sh_audio->sample_format;
-    switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
-    case AV_SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
-    case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
-    case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
-    case AV_SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
-    default:
-        mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
-        sample_format = AF_FORMAT_UNKNOWN;
-    }
-
+    struct priv *priv = sh_audio->context;
+    int sample_format        =
+        af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
     bool broken_srate        = false;
     int samplerate           = lavc_context->sample_rate;
     int container_samplerate = sh_audio->container_out_samplerate;
@@ -166,10 +170,20 @@ static int setup_format(sh_audio_t *sh_audio,
     else if (container_samplerate)
         samplerate = container_samplerate;
 
-    if (lavc_context->channels != sh_audio->channels ||
+    struct mp_chmap lavc_chmap;
+    mp_chmap_from_lavc(&lavc_chmap, lavc_context->channel_layout);
+    // No channel layout or layout disagrees with channel count
+    if (lavc_chmap.num != lavc_context->channels)
+        mp_chmap_from_channels(&lavc_chmap, lavc_context->channels);
+    if (priv->force_channel_map) {
+        if (lavc_chmap.num == sh_audio->channels.num)
+            lavc_chmap = sh_audio->channels;
+    }
+
+    if (!mp_chmap_equals(&lavc_chmap, &sh_audio->channels) ||
         samplerate != sh_audio->samplerate ||
         sample_format != sh_audio->sample_format) {
-        sh_audio->channels = lavc_context->channels;
+        sh_audio->channels = lavc_chmap;
         sh_audio->samplerate = samplerate;
         sh_audio->sample_format = sample_format;
         sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
@@ -198,41 +212,59 @@ static void set_from_wf(AVCodecContext *avctx, WAVEFORMATEX *wf)
 
 static int init(sh_audio_t *sh_audio, const char *decoder)
 {
-    struct MPOpts *opts = sh_audio->opts;
+    struct MPOpts *mpopts = sh_audio->opts;
+    struct ad_lavc_param *opts = &mpopts->ad_lavc_param;
     AVCodecContext *lavc_context;
     AVCodec *lavc_codec;
 
+    struct priv *ctx = talloc_zero(NULL, struct priv);
+    sh_audio->context = ctx;
+
     if (sh_audio->wf && strcmp(decoder, "pcm") == 0) {
         decoder = find_pcm_decoder(tag_map, sh_audio->format,
                                    sh_audio->wf->wBitsPerSample);
     } else if (sh_audio->wf && strcmp(decoder, "mp-pcm") == 0) {
         decoder = find_pcm_decoder(af_map, sh_audio->format, 0);
+        ctx->force_channel_map = true;
     }
 
     lavc_codec = avcodec_find_decoder_by_name(decoder);
     if (!lavc_codec) {
         mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
                 "Cannot find codec '%s' in libavcodec...\n", decoder);
+        uninit(sh_audio);
         return 0;
     }
 
-    struct priv *ctx = talloc_zero(NULL, struct priv);
-    sh_audio->context = ctx;
     lavc_context = avcodec_alloc_context3(lavc_codec);
     ctx->avctx = lavc_context;
     ctx->avframe = avcodec_alloc_frame();
     lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
     lavc_context->codec_id = lavc_codec->id;
 
-    lavc_context->request_channels = opts->audio_output_channels;
+    if (opts->downmix) {
+        lavc_context->request_channels = mpopts->audio_output_channels.num;
+        lavc_context->request_channel_layout =
+            mp_chmap_to_lavc(&mpopts->audio_output_channels);
+    }
 
     // Always try to set - option only exists for AC3 at the moment
-    av_opt_set_double(lavc_context, "drc_scale", opts->drc_level,
+    av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
                       AV_OPT_SEARCH_CHILDREN);
 
+    if (opts->avopt) {
+        if (parse_avopts(lavc_context, opts->avopt) < 0) {
+            mp_msg(MSGT_DECVIDEO, MSGL_ERR,
+                   "ad_lavc: setting AVOptions '%s' failed.\n", opts->avopt);
+            uninit(sh_audio);
+            return 0;
+        }
+    }
+
     lavc_context->codec_tag = sh_audio->format;
     lavc_context->sample_rate = sh_audio->samplerate;
     lavc_context->bit_rate = sh_audio->i_bps * 8;
+    lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
 
     if (sh_audio->wf)
         set_from_wf(lavc_context, sh_audio->wf);
@@ -279,13 +311,9 @@ static int init(sh_audio_t *sh_audio, const char *decoder)
     if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
         sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
 
-    switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
-    case AV_SAMPLE_FMT_U8:
-    case AV_SAMPLE_FMT_S16:
-    case AV_SAMPLE_FMT_S32:
-    case AV_SAMPLE_FMT_FLT:
-        break;
-    default:
+    int af_sample_fmt =
+        af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
+    if (af_sample_fmt == AF_FORMAT_UNKNOWN) {
         uninit(sh_audio);
         return 0;
     }
@@ -442,13 +470,6 @@ static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
         memcpy(buf, priv->output, size);
         priv->output += size;
         priv->output_left -= size;
-        if (avctx->channels >= 5) {
-            int samplesize = av_get_bytes_per_sample(avctx->sample_fmt);
-            reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
-                                AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
-                                avctx->channels,
-                                size / samplesize, samplesize);
-        }
         if (len < 0)
             len = size;
         else
diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
index c17edc9197..8699013acf 100644
--- a/audio/decode/ad_mpg123.c
+++ b/audio/decode/ad_mpg123.c
@@ -358,7 +358,7 @@ static int init(sh_audio_t *sh, const char *decoder)
         con->mean_count = 0;
 #endif
         con->vbr = (finfo.vbr != MPG123_CBR);
-        sh->channels   = channels;
+        mp_chmap_from_channels(&sh->channels, channels);
         sh->samplerate = rate;
         /* Without external force, mpg123 will always choose signed encoding,
          * and non-16-bit only on builds that don't support it.
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 2101721430..1314110062 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -148,19 +148,20 @@ static int init(sh_audio_t *sh, const char *decoder)
     }
     sh->ds->buffer_pos -= in_size;
 
+    int num_channels = 0;
     switch (lavf_ctx->streams[0]->codec->codec_id) {
     case AV_CODEC_ID_AAC:
         spdif_ctx->iec61937_packet_size = 16384;
         sh->sample_format               = AF_FORMAT_IEC61937_LE;
         sh->samplerate                  = srate;
-        sh->channels                    = 2;
+        num_channels                    = 2;
         sh->i_bps                       = bps;
         break;
     case AV_CODEC_ID_AC3:
         spdif_ctx->iec61937_packet_size = 6144;
         sh->sample_format               = AF_FORMAT_AC3_LE;
         sh->samplerate                  = srate;
-        sh->channels                    = 2;
+        num_channels                    = 2;
         sh->i_bps                       = bps;
         break;
     case AV_CODEC_ID_DTS:
@@ -175,13 +176,13 @@ static int init(sh_audio_t *sh, const char *decoder)
             spdif_ctx->iec61937_packet_size = 32768;
             sh->sample_format               = AF_FORMAT_IEC61937_LE;
             sh->samplerate                  = 192000; // DTS core require 48000
-            sh->channels                    = 2*4;
+            num_channels                    = 2*4;
             sh->i_bps                       = bps;
         } else {
             spdif_ctx->iec61937_packet_size = 32768;
             sh->sample_format               = AF_FORMAT_AC3_LE;
             sh->samplerate                  = srate;
-            sh->channels                    = 2;
+            num_channels                    = 2;
             sh->i_bps                       = bps;
         }
         break;
@@ -189,26 +190,28 @@ static int init(sh_audio_t *sh, const char *decoder)
         spdif_ctx->iec61937_packet_size = 24576;
         sh->sample_format               = AF_FORMAT_IEC61937_LE;
         sh->samplerate                  = 192000;
-        sh->channels                    = 2;
+        num_channels                    = 2;
         sh->i_bps                       = bps;
         break;
     case AV_CODEC_ID_MP3:
         spdif_ctx->iec61937_packet_size = 4608;
         sh->sample_format               = AF_FORMAT_MPEG2;
         sh->samplerate                  = srate;
-        sh->channels                    = 2;
+        num_channels                    = 2;
         sh->i_bps                       = bps;
         break;
     case AV_CODEC_ID_TRUEHD:
         spdif_ctx->iec61937_packet_size = 61440;
         sh->sample_format               = AF_FORMAT_IEC61937_LE;
         sh->samplerate                  = 192000;
-        sh->channels                    = 8;
+        num_channels                    = 8;
         sh->i_bps                       = bps;
         break;
     default:
         break;
     }
+    if (num_channels)
+        mp_chmap_from_channels(&sh->channels, num_channels);
 
     return 1;
 
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index cac33a9a31..dc461b81e3 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -41,22 +41,14 @@
 
 int fakemono = 0;
 
-struct af_cfg af_cfg = {1, NULL}; // Configuration for audio filters
+struct af_cfg af_cfg = {0}; // Configuration for audio filters
 
 static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
 {
     assert(!sh_audio->initialized);
     resync_audio_stream(sh_audio);
-    sh_audio->samplesize = 2;
-    sh_audio->sample_format = AF_FORMAT_S16_NE;
-    if ((af_cfg.force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT) {
-        int fmt = AF_FORMAT_FLOAT_NE;
-        if (sh_audio->ad_driver->control(sh_audio, ADCTRL_QUERY_FORMAT,
-                                         &fmt) == CONTROL_TRUE) {
-            sh_audio->sample_format = fmt;
-            sh_audio->samplesize = 4;
-        }
-    }
+    sh_audio->samplesize = 4;
+    sh_audio->sample_format = AF_FORMAT_FLOAT_NE;
     sh_audio->audio_out_minsize = 8192; // default, preinit() may change it
     if (!sh_audio->ad_driver->preinit(sh_audio)) {
         mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder preinit failed.\n");
@@ -94,7 +86,7 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
 
     sh_audio->initialized = 1;
 
-    if (!sh_audio->channels || !sh_audio->samplerate) {
+    if (mp_chmap_is_empty(&sh_audio->channels) || !sh_audio->samplerate) {
         mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder did not specify "
                 "audio format!\n");
         uninit_audio(sh_audio); // free buffers
@@ -102,7 +94,7 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
     }
 
     if (!sh_audio->o_bps)
-        sh_audio->o_bps = sh_audio->channels * sh_audio->samplerate
+        sh_audio->o_bps = sh_audio->channels.num * sh_audio->samplerate
                           * sh_audio->samplesize;
     return 1;
 }
@@ -168,14 +160,14 @@ int init_best_audio_codec(sh_audio_t *sh_audio, char *audio_decoders)
                sh_audio->gsh->decoder_desc);
         mp_msg(MSGT_DECAUDIO, MSGL_V,
                "AUDIO: %d Hz, %d ch, %s, %3.1f kbit/%3.2f%% (ratio: %d->%d)\n",
-               sh_audio->samplerate, sh_audio->channels,
+               sh_audio->samplerate, sh_audio->channels.num,
                af_fmt2str_short(sh_audio->sample_format),
                sh_audio->i_bps * 8 * 0.001,
                ((float) sh_audio->i_bps / sh_audio->o_bps) * 100.0,
                sh_audio->i_bps, sh_audio->o_bps);
         mp_msg(MSGT_IDENTIFY, MSGL_INFO,
                "ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
-               sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels);
+               sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels.num);
     } else {
         mp_msg(MSGT_DECAUDIO, MSGL_ERR,
                "Failed to initialize an audio decoder for codec '%s'.\n",
@@ -191,7 +183,7 @@ void uninit_audio(sh_audio_t *sh_audio)
     if (sh_audio->afilter) {
         mp_msg(MSGT_DECAUDIO, MSGL_V, "Uninit audio filters...\n");
         af_uninit(sh_audio->afilter);
-        free(sh_audio->afilter);
+        af_destroy(sh_audio->afilter);
         sh_audio->afilter = NULL;
     }
     if (sh_audio->initialized) {
@@ -207,43 +199,41 @@ void uninit_audio(sh_audio_t *sh_audio)
 
 
 int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
-                       int *out_samplerate, int *out_channels, int *out_format)
+                       int *out_samplerate, struct mp_chmap *out_channels,
+                       int *out_format)
 {
     struct af_stream *afs = sh_audio->afilter;
-    if (!afs) {
-        afs = calloc(1, sizeof(struct af_stream));
-        afs->opts = sh_audio->opts;
-    }
+    if (!afs)
+        afs = af_new(sh_audio->opts);
     // input format: same as codec's output format:
-    afs->input.rate   = in_samplerate;
-    afs->input.nch    = sh_audio->channels;
-    afs->input.format = sh_audio->sample_format;
-    af_fix_parameters(&(afs->input));
+    afs->input.rate = in_samplerate;
+    mp_audio_set_channels(&afs->input, &sh_audio->channels);
+    mp_audio_set_format(&afs->input, sh_audio->sample_format);
 
     // output format: same as ao driver's input format (if missing, fallback to input)
-    afs->output.rate   = *out_samplerate;
-    afs->output.nch    = *out_channels;
-    afs->output.format = *out_format;
-    af_fix_parameters(&(afs->output));
+    afs->output.rate = *out_samplerate;
+    mp_audio_set_channels(&afs->output, out_channels);
+    mp_audio_set_format(&afs->output, *out_format);
 
     // filter config:
     memcpy(&afs->cfg, &af_cfg, sizeof(struct af_cfg));
 
+    char *s_from = mp_audio_config_to_str(&afs->input);
+    char *s_to = mp_audio_config_to_str(&afs->output);
     mp_tmsg(MSGT_DECAUDIO, MSGL_V,
-            "Building audio filter chain for %dHz/%dch/%s -> %dHz/%dch/%s...\n",
-            afs->input.rate, afs->input.nch,
-            af_fmt2str_short(afs->input.format), afs->output.rate,
-            afs->output.nch, af_fmt2str_short(afs->output.format));
+            "Building audio filter chain for %s -> %s...\n", s_from, s_to);
+    talloc_free(s_from);
+    talloc_free(s_to);
 
     // let's autoprobe it!
     if (0 != af_init(afs)) {
         sh_audio->afilter = NULL;
-        free(afs);
+        af_destroy(afs);
         return 0;   // failed :(
     }
 
     *out_samplerate = afs->output.rate;
-    *out_channels = afs->output.nch;
+    *out_channels = afs->output.channels;
     *out_format = afs->output.format;
 
     // ok!
@@ -270,7 +260,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
 
     // Decode more bytes if needed
     int old_samplerate = sh->samplerate;
-    int old_channels = sh->channels;
+    struct mp_chmap old_channels = sh->channels;
     int old_sample_format = sh->sample_format;
     while (sh->a_buffer_len < len) {
         unsigned char *buf = sh->a_buffer + sh->a_buffer_len;
@@ -278,7 +268,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
         int maxlen = sh->a_buffer_size - sh->a_buffer_len;
         int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
         int format_change = sh->samplerate != old_samplerate
-                            || sh->channels != old_channels
+                            || !mp_chmap_equals(&sh->channels, &old_channels)
                             || sh->sample_format != old_sample_format;
         if (ret <= 0 || format_change) {
             error = format_change ? -2 : -1;
@@ -294,10 +284,10 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
         .audio = sh->a_buffer,
         .len = len,
         .rate = sh->samplerate,
-        .nch = sh->channels,
-        .format = sh->sample_format
     };
-    af_fix_parameters(&filter_input);
+    mp_audio_set_format(&filter_input, sh->sample_format);
+    mp_audio_set_channels(&filter_input, &sh->channels);
+
     struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
     if (!filter_output)
         return -1;
@@ -325,7 +315,7 @@ int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
     // Indicates that a filter seems to be buffering large amounts of data
     int huge_filter_buffer = 0;
     // Decoded audio must be cut at boundaries of this many bytes
-    int unitsize = sh_audio->channels * sh_audio->samplesize * 16;
+    int unitsize = sh_audio->channels.num * sh_audio->samplesize * 16;
 
     /* Filter output size will be about filter_multiplier times input size.
      * If some filter buffers audio in big blocks this might only hold
diff --git a/audio/decode/dec_audio.h b/audio/decode/dec_audio.h
index 9fa6aad8dd..b46f4282fb 100644
--- a/audio/decode/dec_audio.h
+++ b/audio/decode/dec_audio.h
@@ -19,6 +19,7 @@
 #ifndef MPLAYER_DEC_AUDIO_H
 #define MPLAYER_DEC_AUDIO_H
 
+#include "audio/chmap.h"
 #include "demux/stheader.h"
 
 struct bstr;
@@ -33,6 +34,7 @@ void skip_audio_frame(sh_audio_t *sh_audio);
 void uninit_audio(sh_audio_t *sh_audio);
 
 int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
-                       int *out_samplerate, int *out_channels, int *out_format);
+                       int *out_samplerate, struct mp_chmap *out_channels,
+                       int *out_format);
 
 #endif /* MPLAYER_DEC_AUDIO_H */
diff --git a/audio/filter/af.c b/audio/filter/af.c
index ad43e5fca7..137e7cc407 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -20,6 +20,7 @@
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
+#include <assert.h>
 
 #include "af.h"
 
@@ -28,6 +29,7 @@ extern struct af_info af_info_dummy;
 extern struct af_info af_info_delay;
 extern struct af_info af_info_channels;
 extern struct af_info af_info_format;
+extern struct af_info af_info_force;
 extern struct af_info af_info_volume;
 extern struct af_info af_info_equalizer;
 extern struct af_info af_info_pan;
@@ -47,232 +49,389 @@ extern struct af_info af_info_karaoke;
 extern struct af_info af_info_scaletempo;
 extern struct af_info af_info_bs2b;
 
-static struct af_info* filter_list[]={
-   &af_info_dummy,
-   &af_info_delay,
-   &af_info_channels,
-   &af_info_format,
-   &af_info_volume,
-   &af_info_equalizer,
-   &af_info_pan,
-   &af_info_surround,
-   &af_info_sub,
+static struct af_info* filter_list[] = {
+    &af_info_dummy,
+    &af_info_delay,
+    &af_info_channels,
+    &af_info_force,
+    &af_info_volume,
+    &af_info_equalizer,
+    &af_info_pan,
+    &af_info_surround,
+    &af_info_sub,
 #ifdef HAVE_SYS_MMAN_H
-   &af_info_export,
+    &af_info_export,
 #endif
-   &af_info_drc,
-   &af_info_extrastereo,
-   &af_info_lavcac3enc,
-   &af_info_lavrresample,
-   &af_info_sweep,
-   &af_info_hrtf,
+    &af_info_drc,
+    &af_info_extrastereo,
+    &af_info_lavcac3enc,
+    &af_info_lavrresample,
+    &af_info_sweep,
+    &af_info_hrtf,
 #ifdef CONFIG_LADSPA
-   &af_info_ladspa,
+    &af_info_ladspa,
 #endif
-   &af_info_center,
-   &af_info_sinesuppress,
-   &af_info_karaoke,
-   &af_info_scaletempo,
+    &af_info_center,
+    &af_info_sinesuppress,
+    &af_info_karaoke,
+    &af_info_scaletempo,
 #ifdef CONFIG_LIBBS2B
-   &af_info_bs2b,
+    &af_info_bs2b,
 #endif
-   NULL
+    // Must come last, because it's the fallback format conversion filter
+    &af_info_format,
+    NULL
 };
 
-// CPU speed
-int* af_cpu_speed = NULL;
+static bool af_config_equals(struct mp_audio *a, struct mp_audio *b)
+{
+    return a->format == b->format
+        && mp_chmap_equals(&a->channels, &b->channels)
+        && a->rate   == b->rate;
+}
+
+static void af_copy_unset_fields(struct mp_audio *dst, struct mp_audio *src)
+{
+    if (dst->format == AF_FORMAT_UNKNOWN)
+        mp_audio_set_format(dst, src->format);
+    if (dst->nch == 0)
+        mp_audio_set_channels(dst, &src->channels);
+    if (dst->rate == 0)
+        dst->rate = src->rate;
+}
+
+static int input_control(struct af_instance* af, int cmd, void* arg)
+{
+    switch (cmd) {
+    case AF_CONTROL_REINIT:
+        assert(arg == &((struct af_stream *)af->setup)->input);
+        return AF_OK;
+    }
+    return AF_UNKNOWN;
+}
+
+static int output_control(struct af_instance* af, int cmd, void* arg)
+{
+    struct af_stream *s = af->setup;
+    struct mp_audio *output = &s->output;
+    struct mp_audio *filter_output = &s->filter_output;
+
+    switch (cmd) {
+    case AF_CONTROL_REINIT: {
+        struct mp_audio *in = arg;
+        struct mp_audio orig_in = *in;
+
+        *filter_output = *output;
+        af_copy_unset_fields(filter_output, in);
+        *in = *filter_output;
+        return af_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
+    }
+    }
+    return AF_UNKNOWN;
+}
+
+static struct mp_audio *dummy_play(struct af_instance* af, struct mp_audio* data)
+{
+    return data;
+}
 
 /* Find a filter in the static list of filters using it's name. This
    function is used internally */
-static struct af_info* af_find(char*name)
+static struct af_info *af_find(char *name)
 {
-  int i=0;
-  while(filter_list[i]){
-    if(!strcmp(filter_list[i]->name,name))
-      return filter_list[i];
-    i++;
-  }
-  mp_msg(MSGT_AFILTER, MSGL_ERR, "Couldn't find audio filter '%s'\n",name);
-  return NULL;
+    int i = 0;
+    while (filter_list[i]) {
+        if (!strcmp(filter_list[i]->name, name))
+            return filter_list[i];
+        i++;
+    }
+    mp_msg(MSGT_AFILTER, MSGL_ERR, "Couldn't find audio filter '%s'\n", name);
+    return NULL;
 }
 
 /* Find filter in the dynamic filter list using it's name This
    function is used for finding already initialized filters */
-struct af_instance* af_get(struct af_stream* s, char* name)
-{
-  struct af_instance* af=s->first;
-  // Find the filter
-  while(af != NULL){
-    if(!strcmp(af->info->name,name))
-      return af;
-    af=af->next;
-  }
-  return NULL;
-}
-
-/*/ Function for creating a new filter of type name. The name may
-  contain the commandline parameters for the filter */
-static struct af_instance* af_create(struct af_stream* s, const char* name_with_cmd)
-{
-  char* name = strdup(name_with_cmd);
-  char* cmdline = name;
-
-  // Allocate space for the new filter and reset all pointers
-  struct af_instance* new=malloc(sizeof(struct af_instance));
-  if (!name || !new) {
-    mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Could not allocate memory\n");
-    goto err_out;
-  }
-  memset(new,0,sizeof(struct af_instance));
-
-  // Check for commandline parameters
-  char *skip = strstr(cmdline, "=");
-  if (skip) {
-      *skip = '\0'; // for name
-      cmdline = skip + 1;
-  } else {
-      cmdline = NULL;
-  }
-
-  // Find filter from name
-  if(NULL == (new->info=af_find(name)))
-    goto err_out;
-
-  /* Make sure that the filter is not already in the list if it is
-     non-reentrant */
-  if(new->info->flags & AF_FLAGS_NOT_REENTRANT){
-    if(af_get(s,name)){
-      mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] There can only be one instance of"
-	     " the filter '%s' in each stream\n",name);
-      goto err_out;
+struct af_instance *af_get(struct af_stream *s, char *name)
+{
+    struct af_instance *af = s->first;
+    // Find the filter
+    while (af != NULL) {
+        if (!strcmp(af->info->name, name))
+            return af;
+        af = af->next;
     }
-  }
+    return NULL;
+}
 
-  mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Adding filter %s \n",name);
+/* Function for creating a new filter of type name.The name may
+contain the commandline parameters for the filter */
+static struct af_instance *af_create(struct af_stream *s,
+                                     const char *name_with_cmd)
+{
+    char *name = strdup(name_with_cmd);
+    char *cmdline = name;
 
-  // Initialize the new filter
-  if(AF_OK == new->info->open(new) &&
-     AF_ERROR < new->control(new,AF_CONTROL_POST_CREATE,&s->cfg)){
-    if(cmdline){
-      if(AF_ERROR>=new->control(new,AF_CONTROL_COMMAND_LINE,cmdline))
+    // Allocate space for the new filter and reset all pointers
+    struct af_instance *new = malloc(sizeof(struct af_instance));
+    if (!name || !new) {
+        mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Could not allocate memory\n");
         goto err_out;
     }
-    free(name);
-    return new;
-  }
+    memset(new, 0, sizeof(struct af_instance));
+
+    // Check for commandline parameters
+    char *skip = strstr(cmdline, "=");
+    if (skip) {
+        *skip = '\0'; // for name
+        cmdline = skip + 1;
+    } else {
+        cmdline = NULL;
+    }
+
+    // Find filter from name
+    if (NULL == (new->info = af_find(name)))
+        goto err_out;
+
+    /* Make sure that the filter is not already in the list if it is
+       non-reentrant */
+    if (new->info->flags & AF_FLAGS_NOT_REENTRANT) {
+        if (af_get(s, name)) {
+            mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] There can only be one "
+                   "instance of the filter '%s' in each stream\n", name);
+            goto err_out;
+        }
+    }
+
+    mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Adding filter %s \n", name);
+
+    // Initialize the new filter
+    if (AF_OK == new->info->open(new)) {
+        if (cmdline) {
+            if (AF_ERROR >= new->control(new, AF_CONTROL_COMMAND_LINE, cmdline))
+                goto err_out;
+        }
+        free(name);
+        return new;
+    }
 
 err_out:
-  free(new);
-  mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Couldn't create or open audio filter '%s'\n",
-	 name);
-  free(name);
-  return NULL;
+    free(new);
+    mp_msg(MSGT_AFILTER, MSGL_ERR,
+           "[libaf] Couldn't create or open audio filter '%s'\n", name);
+    free(name);
+    return NULL;
 }
 
 /* Create and insert a new filter of type name before the filter in the
    argument. This function can be called during runtime, the return
    value is the new filter */
-static struct af_instance* af_prepend(struct af_stream* s, struct af_instance* af, const char* name)
+static struct af_instance *af_prepend(struct af_stream *s,
+                                      struct af_instance *af,
+                                      const char *name)
 {
-  // Create the new filter and make sure it is OK
-  struct af_instance* new=af_create(s,name);
-  if(!new)
-    return NULL;
-  // Update pointers
-  new->next=af;
-  if(af){
-    new->prev=af->prev;
-    af->prev=new;
-  }
-  else
-    s->last=new;
-  if(new->prev)
-    new->prev->next=new;
-  else
-    s->first=new;
-  return new;
+    if (!af)
+        af = s->last;
+    if (af == s->first)
+        af = s->first->next;
+    // Create the new filter and make sure it is OK
+    struct af_instance *new = af_create(s, name);
+    if (!new)
+        return NULL;
+    // Update pointers
+    new->next = af;
+    new->prev = af->prev;
+    af->prev = new;
+    new->prev->next = new;
+    return new;
 }
 
 /* Create and insert a new filter of type name after the filter in the
    argument. This function can be called during runtime, the return
    value is the new filter */
-static struct af_instance* af_append(struct af_stream* s, struct af_instance* af, const char* name)
+static struct af_instance *af_append(struct af_stream *s,
+                                     struct af_instance *af,
+                                     const char *name)
 {
-  // Create the new filter and make sure it is OK
-  struct af_instance* new=af_create(s,name);
-  if(!new)
-    return NULL;
-  // Update pointers
-  new->prev=af;
-  if(af){
-    new->next=af->next;
-    af->next=new;
-  }
-  else
-    s->first=new;
-  if(new->next)
-    new->next->prev=new;
-  else
-    s->last=new;
-  return new;
+    if (!af)
+        af = s->first;
+    if (af == s->last)
+        af = s->last->prev;
+    // Create the new filter and make sure it is OK
+    struct af_instance *new = af_create(s, name);
+    if (!new)
+        return NULL;
+    // Update pointers
+    new->prev = af;
+    new->next = af->next;
+    af->next = new;
+    new->next->prev = new;
+    return new;
 }
 
 // Uninit and remove the filter "af"
-void af_remove(struct af_stream* s, struct af_instance* af)
+void af_remove(struct af_stream *s, struct af_instance *af)
 {
-  if(!af) return;
+    if (!af)
+        return;
+
+    if (af == s->first || af == s->last)
+        return;
 
-  // Print friendly message
-  mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Removing filter %s \n",af->info->name);
+    // Print friendly message
+    mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Removing filter %s \n",
+           af->info->name);
 
-  // Notify filter before changing anything
-  af->control(af,AF_CONTROL_PRE_DESTROY,0);
+    // Notify filter before changing anything
+    af->control(af, AF_CONTROL_PRE_DESTROY, 0);
 
-  // Detach pointers
-  if(af->prev)
-    af->prev->next=af->next;
-  else
-    s->first=af->next;
-  if(af->next)
-    af->next->prev=af->prev;
-  else
-    s->last=af->prev;
+    // Detach pointers
+    af->prev->next = af->next;
+    af->next->prev = af->prev;
 
-  // Uninitialize af and free memory
-  af->uninit(af);
-  free(af);
+    af->uninit(af);
+    free(af);
 }
 
-static void print_fmt(struct mp_audio *d)
+static void remove_auto_inserted_filters(struct af_stream *s)
 {
-    if (d) {
-        mp_msg(MSGT_AFILTER, MSGL_V, "%dHz/%dch/%s", d->rate, d->nch,
-               af_fmt2str_short(d->format));
-    } else {
-        mp_msg(MSGT_AFILTER, MSGL_V, "(?)");
+repeat:
+    for (struct af_instance *af = s->first; af; af = af->next) {
+        if (af->auto_inserted) {
+            af_remove(s, af);
+            goto repeat;
+        }
     }
 }
 
-static void af_print_filter_chain(struct af_stream* s)
+static void af_print_filter_chain(struct af_stream *s, struct af_instance *at,
+                                  int msg_level)
 {
-    mp_msg(MSGT_AFILTER, MSGL_V, "Audio filter chain:\n");
-
-    mp_msg(MSGT_AFILTER, MSGL_V, "  [in] ");
-    print_fmt(&s->input);
-    mp_msg(MSGT_AFILTER, MSGL_V, "\n");
+    mp_msg(MSGT_AFILTER, msg_level, "Audio filter chain:\n");
 
     struct af_instance *af = s->first;
     while (af) {
-        mp_msg(MSGT_AFILTER, MSGL_V, "  [%s] ", af->info->name);
-        print_fmt(af->data);
-        mp_msg(MSGT_AFILTER, MSGL_V, "\n");
+        mp_msg(MSGT_AFILTER, msg_level, "  [%s] ", af->info->name);
+        if (af->data) {
+            char *info = mp_audio_config_to_str(af->data);
+            mp_msg(MSGT_AFILTER, msg_level, "%s", info);
+            talloc_free(info);
+        }
+        if (af == at)
+            mp_msg(MSGT_AFILTER, msg_level, " <-");
+        mp_msg(MSGT_AFILTER, msg_level, "\n");
 
         af = af->next;
     }
 
-    mp_msg(MSGT_AFILTER, MSGL_V, "  [out] ");
-    print_fmt(&s->output);
-    mp_msg(MSGT_AFILTER, MSGL_V, "\n");
+    mp_msg(MSGT_AFILTER, msg_level, "  [ao] ");
+    char *info = mp_audio_config_to_str(&s->output);
+    mp_msg(MSGT_AFILTER, msg_level, "%s\n", info);
+    talloc_free(info);
+}
+
+static int af_count_filters(struct af_stream *s)
+{
+    int count = 0;
+    for (struct af_instance *af = s->first; af; af = af->next)
+        count++;
+    return count;
+}
+
+static const char *af_find_conversion_filter(int srcfmt, int dstfmt)
+{
+    for (int n = 0; filter_list[n]; n++) {
+        struct af_info *af = filter_list[n];
+        if (af->test_conversion && af->test_conversion(srcfmt, dstfmt))
+            return af->name;
+    }
+    return NULL;
+}
+
+static bool af_is_conversion_filter(struct af_instance *af)
+{
+    return af && af->info->test_conversion != NULL;
+}
+
+// in is what af can take as input - insert a conversion filter if the actual
+// input format doesn't match what af expects.
+// Returns:
+//   AF_OK: must call af_reinit() or equivalent, format matches
+//   AF_FALSE: nothing was changed, format matches
+//   else: error
+static int af_fix_format_conversion(struct af_stream *s,
+                                    struct af_instance **p_af,
+                                    struct mp_audio in)
+{
+    int rv;
+    struct af_instance *af = *p_af;
+    struct af_instance *prev = af->prev;
+    struct mp_audio actual = *prev->data;
+    if (actual.format == in.format)
+        return AF_FALSE;
+    if (prev->control(prev, AF_CONTROL_FORMAT_FMT, &in.format) == AF_OK) {
+        *p_af = prev;
+        return AF_OK;
+    }
+    const char *filter = af_find_conversion_filter(actual.format, in.format);
+    if (!filter)
+        return AF_ERROR;
+    struct af_instance *new = af_prepend(s, af, filter);
+    if (new == NULL)
+        return AF_ERROR;
+    new->auto_inserted = true;
+    if (AF_OK != (rv = new->control(new, AF_CONTROL_FORMAT_FMT, &in.format)))
+        return rv;
+    *p_af = new;
+    return AF_OK;
+}
+
+// same as af_fix_format_conversion - only wrt. channels
+static int af_fix_channels(struct af_stream *s, struct af_instance **p_af,
+                           struct mp_audio in)
+{
+    int rv;
+    struct af_instance *af = *p_af;
+    struct af_instance *prev = af->prev;
+    struct mp_audio actual = *prev->data;
+    if (mp_chmap_equals(&actual.channels, &in.channels))
+        return AF_FALSE;
+    if (prev->control(prev, AF_CONTROL_CHANNELS, &in.channels) == AF_OK) {
+        *p_af = prev;
+        return AF_OK;
+    }
+    const char *filter = "lavrresample";
+    struct af_instance *new = af_prepend(s, af, filter);
+    if (new == NULL)
+        return AF_ERROR;
+    new->auto_inserted = true;
+    if (AF_OK != (rv = new->control(new, AF_CONTROL_CHANNELS, &in.channels)))
+        return rv;
+    *p_af = new;
+    return AF_OK;
+}
+
+static int af_fix_rate(struct af_stream *s, struct af_instance **p_af,
+                       struct mp_audio in)
+{
+    int rv;
+    struct af_instance *af = *p_af;
+    struct af_instance *prev = af->prev;
+    struct mp_audio actual = *prev->data;
+    if (actual.rate == in.rate)
+        return AF_FALSE;
+    if (prev->control(prev, AF_CONTROL_RESAMPLE_RATE, &in.rate) == AF_OK) {
+        *p_af = prev;
+        return AF_OK;
+    }
+    const char *filter = "lavrresample";
+    struct af_instance *new = af_prepend(s, af, filter);
+    if (new == NULL)
+        return AF_ERROR;
+    new->auto_inserted = true;
+    if (AF_OK != (rv = new->control(new, AF_CONTROL_RESAMPLE_RATE, &in.rate)))
+        return rv;
+    *p_af = new;
+    return AF_OK;
 }
 
 // Warning:
@@ -280,186 +439,128 @@ static void af_print_filter_chain(struct af_stream* s)
 // state (for example, format filters that were tentatively inserted stay
 // inserted).
 // In that case, you should always rebuild the filter chain, or abort.
-int af_reinit(struct af_stream* s, struct af_instance* af)
-{
-  do{
-    struct mp_audio in; // Format of the input to current filter
-    int rv=0; // Return value
-
-    // Check if there are any filters left in the list
-    if(NULL == af){
-      if(!(af=af_append(s,s->first,"dummy")))
-	return AF_UNKNOWN;
-      else
-	return AF_ERROR;
+// Also, note that for complete reinit, fixup_output_format() may have to be
+// called after this function.
+int af_reinit(struct af_stream *s)
+{
+    // Start with the second filter, as the first filter is the special input
+    // filter which needs no initialization.
+    struct af_instance *af = s->first->next;
+    int max_retry = af_count_filters(s) * 4; // up to 4 retries per filter
+    int retry = 0;
+    while (af) {
+        if (retry >= max_retry)
+            goto negotiate_error;
+
+        // Check if this is the first filter
+        struct mp_audio in = *af->prev->data;
+        // Reset just in case...
+        in.audio = NULL;
+        in.len = 0;
+
+        int rv = af->control(af, AF_CONTROL_REINIT, &in);
+        switch (rv) {
+        case AF_OK:
+            af = af->next;
+            break;
+        case AF_FALSE: { // Configuration filter is needed
+            if (af_fix_channels(s, &af, in) == AF_OK) {
+                retry++;
+                continue;
+            }
+            if (af_fix_rate(s, &af, in) == AF_OK) {
+                retry++;
+                continue;
+            }
+            // Do this last, to prevent "format->lavrresample" being added to
+            // the filter chain when output formats not supported by
+            // af_lavrresample are in use.
+            if (af_fix_format_conversion(s, &af, in) == AF_OK) {
+                retry++;
+                continue;
+            }
+            goto negotiate_error;
+        }
+        case AF_DETACH: { // Filter is redundant and wants to be unloaded
+            struct af_instance *aft = af->prev; // never NULL
+            af_remove(s, af);
+            af = aft->next;
+            break;
+        }
+        default:
+            mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Reinitialization did not "
+                   "work, audio filter '%s' returned error code %i\n",
+                   af->info->name, rv);
+            af_print_filter_chain(s, af, MSGL_ERR);
+            return AF_ERROR;
+        }
     }
 
-    // Check if this is the first filter
-    if(!af->prev)
-      memcpy(&in,&(s->input),sizeof(struct mp_audio));
-    else
-      memcpy(&in,af->prev->data,sizeof(struct mp_audio));
-    // Reset just in case...
-    in.audio=NULL;
-    in.len=0;
-
-    rv = af->control(af,AF_CONTROL_REINIT,&in);
-    switch(rv){
-    case AF_OK:
-	af = af->next;
-      break;
-    case AF_FALSE:{ // Configuration filter is needed
-      // Do auto insertion only if force is not specified
-      if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
-	struct af_instance* new = NULL;
-	// Insert channels filter
-	if((af->prev?af->prev->data->nch:s->input.nch) != in.nch){
-	  // Create channels filter
-	  if(NULL == (new = af_prepend(s,af,"channels")))
-	    return AF_ERROR;
-	  // Set number of output channels
-	  if(AF_OK != (rv = new->control(new,AF_CONTROL_CHANNELS,&in.nch)))
-	    return rv;
-	  // Initialize channels filter
-	  if(!new->prev)
-	    memcpy(&in,&(s->input),sizeof(struct mp_audio));
-	  else
-	    memcpy(&in,new->prev->data,sizeof(struct mp_audio));
-	  if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
-	    return rv;
-	}
-	// Insert format filter
-	if((af->prev?af->prev->data->format:s->input.format) != in.format){
-	  // Create format filter
-	  if(NULL == (new = af_prepend(s,af,"format")))
-	    return AF_ERROR;
-	  // Set output bits per sample
-	  in.format |= af_bits2fmt(in.bps*8);
-	  if(AF_OK != (rv = new->control(new,AF_CONTROL_FORMAT_FMT,&in.format)))
-	    return rv;
-	  // Initialize format filter
-	  if(!new->prev)
-	    memcpy(&in,&(s->input),sizeof(struct mp_audio));
-	  else
-	    memcpy(&in,new->prev->data,sizeof(struct mp_audio));
-	  if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
-	    return rv;
-	}
-	if(!new){ // Should _never_ happen
-	  mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to correct audio format. "
-		 "This error should never occur, please send a bug report.\n");
-	  return AF_ERROR;
-	}
-	af=new->next;
-      }
-      else {
-        mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Automatic filter insertion disabled "
-               "but formats do not match. Giving up.\n");
-        return AF_ERROR;
-      }
-      break;
-    }
-    case AF_DETACH:{ // Filter is redundant and wants to be unloaded
-      // Do auto remove only if force is not specified
-      if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
-	struct af_instance* aft=af->prev;
-	af_remove(s,af);
-	if(aft)
-	  af=aft->next;
-	else
-	  af=s->first; // Restart configuration
-      }
-      break;
-    }
-    default:
-      mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Reinitialization did not work, audio"
-	     " filter '%s' returned error code %i\n",af->info->name,rv);
-      return AF_ERROR;
-    }
-  }while(af);
+    af_print_filter_chain(s, NULL, MSGL_V);
 
-  af_print_filter_chain(s);
+    return AF_OK;
 
-  return AF_OK;
+negotiate_error:
+    mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to correct audio format. "
+           "This error should never occur, please send a bug report.\n");
+    af_print_filter_chain(s, af, MSGL_ERR);
+    return AF_ERROR;
 }
 
 // Uninit and remove all filters
-void af_uninit(struct af_stream* s)
+void af_uninit(struct af_stream *s)
 {
-  while(s->first)
-    af_remove(s,s->first);
+    while (s->first->next && s->first->next != s->last)
+        af_remove(s, s->first->next);
 }
 
-/**
- * Extend the filter chain so we get the required output format at the end.
- * \return AF_ERROR on error, AF_OK if successful.
- */
-static int fixup_output_format(struct af_stream* s)
-{
-    struct af_instance* af = NULL;
-    // Check number of output channels fix if not OK
-    // If needed always inserted last -> easy to screw up other filters
-    if(s->output.nch && s->last->data->nch!=s->output.nch){
-      if(!strcmp(s->last->info->name,"format"))
-	af = af_prepend(s,s->last,"channels");
-      else
-	af = af_append(s,s->last,"channels");
-      // Init the new filter
-      if(!af || (AF_OK != af->control(af,AF_CONTROL_CHANNELS,&(s->output.nch))))
-	return AF_ERROR;
-      if(AF_OK != af_reinit(s,af))
-	return AF_ERROR;
-    }
-
-    // Check output format fix if not OK
-    if(s->output.format != AF_FORMAT_UNKNOWN &&
-		s->last->data->format != s->output.format){
-      if(strcmp(s->last->info->name,"format"))
-	af = af_append(s,s->last,"format");
-      else
-	af = s->last;
-      // Init the new filter
-      s->output.format |= af_bits2fmt(s->output.bps*8);
-      if(!af || (AF_OK != af->control(af,AF_CONTROL_FORMAT_FMT,&(s->output.format))))
-	return AF_ERROR;
-      if(AF_OK != af_reinit(s,af))
-	return AF_ERROR;
-    }
+struct af_stream *af_new(struct MPOpts *opts)
+{
+    struct af_stream *s = talloc_zero(NULL, struct af_stream);
+    static struct af_info in = { .name = "in" };
+    s->first = talloc(s, struct af_instance);
+    *s->first = (struct af_instance) {
+        .info = &in,
+        .control = input_control,
+        .play = dummy_play,
+        .setup = s,
+        .data = &s->input,
+        .mul = 1.0,
+    };
+    static struct af_info out = { .name = "out" };
+    s->last = talloc(s, struct af_instance);
+    *s->last = (struct af_instance) {
+        .info = &out,
+        .control = output_control,
+        .play = dummy_play,
+        .setup = s,
+        .data = &s->filter_output,
+        .mul = 1.0,
+    };
+    s->first->next = s->last;
+    s->last->prev = s->first;
+    return s;
+}
 
-    // Re init again just in case
-    if(AF_OK != af_reinit(s,s->first))
-      return AF_ERROR;
-
-    if (s->output.format == AF_FORMAT_UNKNOWN)
-	s->output.format = s->last->data->format;
-    if (!s->output.nch) s->output.nch = s->last->data->nch;
-    if (!s->output.rate) s->output.rate = s->last->data->rate;
-    if((s->last->data->format != s->output.format) ||
-       (s->last->data->nch    != s->output.nch)    ||
-       (s->last->data->rate   != s->output.rate))  {
-      return AF_ERROR;
-    }
-    return AF_OK;
+void af_destroy(struct af_stream *s)
+{
+    af_uninit(s);
+    talloc_free(s);
 }
 
-/**
- * Automatic downmix to stereo in case the codec does not implement it.
+/*
+ * Set previously unset fields in s->output to those of the filter chain
+ * output. This is used to make the output format fixed, and even if you insert
+ * new filters or change the input format, the output format won't change.
+ * \return AF_ERROR on error, AF_OK if successful.
  */
-static void af_downmix(struct af_stream* s)
-{
-    static const char * const downmix_strs[AF_NCH + 1] = {
-        /*                FL       FR       RL       RR          FC          LF         AL      AR */
-        [3] = "pan=2:" "0.6:0:" "0:0.6:"                     "0.4:0.4",
-        [4] = "pan=2:" "0.6:0:" "0:0.6:" "0.4:0:"  "0:0.4",
-        [5] = "pan=2:" "0.5:0:" "0:0.5:" "0.2:0:"  "0:0.2:"  "0.3:0.3",
-        [6] = "pan=2:" "0.4:0:" "0:0.4:" "0.2:0:"  "0:0.2:"  "0.3:0.3:"   "0.1:0.1",
-        [7] = "pan=2:" "0.4:0:" "0:0.4:" "0.2:0:"  "0:0.2:"  "0.3:0.3:"              "0.1:0:" "0:0.1",
-        [8] = "pan=2:" "0.4:0:" "0:0.4:" "0.15:0:" "0:0.15:" "0.25:0.25:" "0.1:0.1:" "0.1:0:" "0:0.1",
-    };
-    const char *af_pan_str = downmix_strs[s->input.nch];
+static int fixup_output_format(struct af_stream *s)
+{
+    if (AF_OK != af_reinit(s))
+        return AF_ERROR;
 
-    if (af_pan_str)
-        af_append(s, s->first, af_pan_str);
+    af_copy_unset_fields(&s->output, &s->filter_output);
+    return af_config_equals(&s->output, &s->filter_output) ? AF_OK : AF_ERROR;
 }
 
 /* Initialize the stream "s". This function creates a new filter list
@@ -471,221 +572,176 @@ static void af_downmix(struct af_stream* s)
    If one of the prefered output parameters is 0 the one that needs
    no conversion is used (i.e. the output format in the last filter).
    The return value is 0 if success and -1 if failure */
-int af_init(struct af_stream* s)
-{
-    struct MPOpts *opts = s->opts;
-  int i=0;
-
-  // Sanity check
-  if(!s) return -1;
-
-  // Precaution in case caller is misbehaving
-  s->input.audio  = s->output.audio  = NULL;
-  s->input.len    = s->output.len    = 0;
-
-  // Figure out how fast the machine is
-  if(AF_INIT_AUTO == (AF_INIT_TYPE_MASK & s->cfg.force))
-    s->cfg.force = (s->cfg.force & ~AF_INIT_TYPE_MASK) | AF_INIT_TYPE;
-
-  // Check if this is the first call
-  if(!s->first){
-    // Append a downmix pan filter at the beginning of the chain if needed
-    if (s->input.nch != opts->audio_output_channels
-        && opts->audio_output_channels == 2)
-      af_downmix(s);
-    // Add all filters in the list (if there are any)
-    if (s->cfg.list) {
-      while(s->cfg.list[i]){
-	if(!af_append(s,s->last,s->cfg.list[i++]))
-	  return -1;
-      }
-    }
-  }
-
-  // If we do not have any filters otherwise
-  // add dummy to make automatic format conversion work
-  if (!s->first && !af_append(s, s->first, "dummy"))
-    return -1;
-
-  // Init filters
-  if(AF_OK != af_reinit(s,s->first))
-    return -1;
-
-  // make sure the chain is not empty and valid (e.g. because of AF_DETACH)
-  if (!s->first)
-    if (!af_append(s,s->first,"dummy") || AF_OK != af_reinit(s,s->first))
-      return -1;
-
-  // Check output format
-  if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
-    struct af_instance* af = NULL; // New filter
-    // Check output frequency if not OK fix with resample
-    if(s->output.rate && s->last->data->rate!=s->output.rate){
-      // try to find a filter that can change samplrate
-      af = af_control_any_rev(s, AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
-               &(s->output.rate));
-      if (!af) {
-        char *resampler = "lavrresample";
-	if((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW){
-	  if(!strcmp(s->first->info->name,"format"))
-	    af = af_append(s,s->first,resampler);
-	  else
-	    af = af_prepend(s,s->first,resampler);
-	}
-	else{
-	  if(!strcmp(s->last->info->name,"format"))
-	    af = af_prepend(s,s->last,resampler);
-	  else
-	    af = af_append(s,s->last,resampler);
-	}
-      // Init the new filter
-      if(!af || (AF_OK != af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
-				      &(s->output.rate))))
-	return -1;
-      }
-      if(AF_OK != af_reinit(s,af))
-      	return -1;
+int af_init(struct af_stream *s)
+{
+    int i = 0;
+
+    // Sanity check
+    if (!s)
+        return -1;
+
+    // Precaution in case caller is misbehaving
+    s->input.audio  = s->output.audio  = NULL;
+    s->input.len    = s->output.len    = 0;
+
+    // Check if this is the first call
+    if (s->first->next == s->last) {
+        // Add all filters in the list (if there are any)
+        if (s->cfg.list) {
+            while (s->cfg.list[i]) {
+                if (!af_prepend(s, s->last, s->cfg.list[i++]))
+                    return -1;
+            }
+        }
     }
+
+    remove_auto_inserted_filters(s);
+
+    // Init filters
+    if (AF_OK != af_reinit(s))
+        return -1;
+
     if (AF_OK != fixup_output_format(s)) {
-      // Something is stuffed audio out will not work
-      mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to setup filter system can not"
-	     " meet sound-card demands, please send a bug report. \n");
-      af_uninit(s);
-      return -1;
+        // Something is stuffed audio out will not work
+        mp_msg(
+            MSGT_AFILTER, MSGL_ERR,
+            "[libaf] Unable to setup filter system can not"
+            " meet sound-card demands, please send a bug report. \n");
+        af_uninit(s);
+        return -1;
     }
-  }
-  return 0;
+    return 0;
 }
 
 /* Add filter during execution. This function adds the filter "name"
    to the stream s. The filter will be inserted somewhere nice in the
    list of filters. The return value is a pointer to the new filter,
    If the filter couldn't be added the return value is NULL. */
-struct af_instance* af_add(struct af_stream* s, char* name){
-  struct af_instance* new;
-  // Sanity check
-  if(!s || !s->first || !name)
-    return NULL;
-  // Insert the filter somewhere nice
-  if(!strcmp(s->first->info->name,"format"))
-    new = af_append(s, s->first, name);
-  else
-    new = af_prepend(s, s->first, name);
-  if(!new)
-    return NULL;
-
-  // Reinitalize the filter list
-  if(AF_OK != af_reinit(s, s->first) ||
-     AF_OK != fixup_output_format(s)){
-    while (s->first)
-      af_remove(s, s->first);
-    af_init(s);
-    return NULL;
-  }
-  return new;
+struct af_instance *af_add(struct af_stream *s, char *name)
+{
+    struct af_instance *new;
+    // Sanity check
+    if (!s || !s->first || !name)
+        return NULL;
+    // Insert the filter somewhere nice
+    if (af_is_conversion_filter(s->first->next))
+        new = af_append(s, s->first->next, name);
+    else
+        new = af_prepend(s, s->first->next, name);
+    if (!new)
+        return NULL;
+
+    // Reinitalize the filter list
+    if (AF_OK != af_reinit(s) ||
+        AF_OK != fixup_output_format(s)) {
+        while (s->first)
+            af_remove(s, s->first);
+        af_init(s);
+        return NULL;
+    }
+    return new;
 }
 
 // Filter data chunk through the filters in the list
-struct mp_audio* af_play(struct af_stream* s, struct mp_audio* data)
+struct mp_audio *af_play(struct af_stream *s, struct mp_audio *data)
 {
-  struct af_instance* af=s->first;
-  // Iterate through all filters
-  do{
-    if (data->len <= 0) break;
-    data=af->play(af,data);
-    af=af->next;
-  }while(af && data);
-  return data;
+    struct af_instance *af = s->first;
+    // Iterate through all filters
+    do {
+        if (data->len <= 0)
+            break;
+        data = af->play(af, data);
+        af = af->next;
+    } while (af && data);
+    return data;
 }
 
 /* Calculate the minimum output buffer size for given input data d
  * when using the RESIZE_LOCAL_BUFFER macro. The +t+1 part ensures the
  * value is >= len*mul rounded upwards to whole samples even if the
  * double 'mul' is inexact. */
-int af_lencalc(double mul, struct mp_audio* d)
+int af_lencalc(double mul, struct mp_audio *d)
 {
-  int t = d->bps * d->nch;
-  return d->len * mul + t + 1;
+    int t = d->bps * d->nch;
+    return d->len * mul + t + 1;
 }
 
 // Calculate average ratio of filter output size to input size
-double af_calc_filter_multiplier(struct af_stream* s)
+double af_calc_filter_multiplier(struct af_stream *s)
 {
-  struct af_instance* af=s->first;
-  double mul = 1;
-  // Iterate through all filters and calculate total multiplication factor
-  do{
-      mul *= af->mul;
-      af=af->next;
-  }while(af);
+    struct af_instance *af = s->first;
+    double mul = 1;
+    // Iterate through all filters and calculate total multiplication factor
+    do {
+        mul *= af->mul;
+        af = af->next;
+    } while (af);
 
-  return mul;
+    return mul;
 }
 
 /* Calculate the total delay [bytes output] caused by the filters */
-double af_calc_delay(struct af_stream* s)
+double af_calc_delay(struct af_stream *s)
 {
-  struct af_instance* af=s->first;
-  register double delay = 0.0;
-  // Iterate through all filters
-  while(af){
-    delay += af->delay;
-    delay *= af->mul;
-    af=af->next;
-  }
-  return delay;
+    struct af_instance *af = s->first;
+    register double delay = 0.0;
+    // Iterate through all filters
+    while (af) {
+        delay += af->delay;
+        delay *= af->mul;
+        af = af->next;
+    }
+    return delay;
 }
 
 /* Helper function called by the macro with the same name this
    function should not be called directly */
-int af_resize_local_buffer(struct af_instance* af, struct mp_audio* data)
-{
-  // Calculate new length
-  register int len = af_lencalc(af->mul,data);
-  mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Reallocating memory in module %s, "
-	 "old len = %i, new len = %i\n",af->info->name,af->data->len,len);
-  // If there is a buffer free it
-  free(af->data->audio);
-  // Create new buffer and check that it is OK
-  af->data->audio = malloc(len);
-  if(!af->data->audio){
-    mp_msg(MSGT_AFILTER, MSGL_FATAL, "[libaf] Could not allocate memory \n");
-    return AF_ERROR;
-  }
-  af->data->len=len;
-  return AF_OK;
+int af_resize_local_buffer(struct af_instance *af, struct mp_audio *data)
+{
+    // Calculate new length
+    register int len = af_lencalc(af->mul, data);
+    mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Reallocating memory in module %s, "
+           "old len = %i, new len = %i\n", af->info->name, af->data->len, len);
+    // If there is a buffer free it
+    free(af->data->audio);
+    // Create new buffer and check that it is OK
+    af->data->audio = malloc(len);
+    if (!af->data->audio) {
+        mp_msg(MSGT_AFILTER, MSGL_FATAL, "[libaf] Could not allocate memory \n");
+        return AF_ERROR;
+    }
+    af->data->len = len;
+    return AF_OK;
 }
 
 // documentation in af.h
-struct af_instance *af_control_any_rev (struct af_stream* s, int cmd, void* arg) {
-  int res = AF_UNKNOWN;
-  struct af_instance* filt = s->last;
-  while (filt) {
-    res = filt->control(filt, cmd, arg);
-    if (res == AF_OK)
-      return filt;
-    filt = filt->prev;
-  }
-  return NULL;
-}
-
-void af_help (void) {
-  int i = 0;
-  mp_msg(MSGT_AFILTER, MSGL_INFO, "Available audio filters:\n");
-  while (filter_list[i]) {
-    if (filter_list[i]->comment && filter_list[i]->comment[0])
-      mp_msg(MSGT_AFILTER, MSGL_INFO, "  %-15s: %s (%s)\n", filter_list[i]->name, filter_list[i]->info, filter_list[i]->comment);
-    else
-      mp_msg(MSGT_AFILTER, MSGL_INFO, "  %-15s: %s\n", filter_list[i]->name, filter_list[i]->info);
-    i++;
-  }
+struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg)
+{
+    int res = AF_UNKNOWN;
+    struct af_instance *filt = s->last;
+    while (filt) {
+        res = filt->control(filt, cmd, arg);
+        if (res == AF_OK)
+            return filt;
+        filt = filt->prev;
+    }
+    return NULL;
 }
 
-void af_fix_parameters(struct mp_audio *data)
+void af_help(void)
 {
-    if (data->nch < 0 || data->nch > AF_NCH) {
-      mp_msg(MSGT_AFILTER, MSGL_ERR, "Invalid number of channels %i, assuming 2.\n", data->nch);
-      data->nch = 2;
+    int i = 0;
+    mp_msg(MSGT_AFILTER, MSGL_INFO, "Available audio filters:\n");
+    while (filter_list[i]) {
+        if (filter_list[i]->comment && filter_list[i]->comment[0]) {
+            mp_msg(MSGT_AFILTER, MSGL_INFO, "  %-15s: %s (%s)\n",
+                   filter_list[i]->name, filter_list[i]->info,
+                   filter_list[i]->comment);
+        } else {
+            mp_msg(MSGT_AFILTER, MSGL_INFO, "  %-15s: %s\n",
+                   filter_list[i]->name,
+                   filter_list[i]->info);
+        }
+        i++;
     }
-    data->bps = af_fmt2bits(data->format)/8;
 }
diff --git a/audio/filter/af.h b/audio/filter/af.h
index 31abe1edee..4ccc70792f 100644
--- a/audio/filter/af.h
+++ b/audio/filter/af.h
@@ -20,31 +20,21 @@
 #define MPLAYER_AF_H
 
 #include <stdio.h>
+#include <stdbool.h>
 
 #include "config.h"
 
 #include "core/options.h"
 #include "audio/format.h"
+#include "audio/chmap.h"
+#include "audio/audio.h"
 #include "control.h"
 #include "core/mp_msg.h"
 
 struct af_instance;
 
 // Number of channels
-#ifndef AF_NCH
-#define AF_NCH 8
-#endif
-
-// Audio data chunk
-struct mp_audio {
-    void *audio; // data buffer
-    int len;    // buffer length
-    int rate;   // sample rate
-    int nch;    // number of channels
-    int format; // format
-    int bps;    // bytes per sample
-};
-
+#define AF_NCH MP_NUM_CHANNELS
 
 // Flags used for defining the behavior of an audio filter
 #define AF_FLAGS_REENTRANT      0x00000000
@@ -59,6 +49,7 @@ struct af_info {
     const char *comment;
     const int flags;
     int (*open)(struct af_instance *vf);
+    bool (*test_conversion)(int src_format, int dst_format);
 };
 
 // Linked list of audio filters
@@ -75,29 +66,11 @@ struct af_instance {
                    * corresponding output */
     double mul; /* length multiplier: how much does this instance change
                    the length of the buffer. */
+    bool auto_inserted; // inserted by af.c, such as conversion filters
 };
 
-// Initialization flags
-extern int *af_cpu_speed;
-
-#define AF_INIT_AUTO            0x00000000
-#define AF_INIT_SLOW            0x00000001
-#define AF_INIT_FAST            0x00000002
-#define AF_INIT_FORCE           0x00000003
-#define AF_INIT_TYPE_MASK       0x00000003
-
-#define AF_INIT_INT             0x00000000
-#define AF_INIT_FLOAT           0x00000004
-#define AF_INIT_FORMAT_MASK     0x00000004
-
-// Default init type
-#ifndef AF_INIT_TYPE
-#define AF_INIT_TYPE (af_cpu_speed ? *af_cpu_speed : AF_INIT_SLOW)
-#endif
-
 // Configuration switches
 struct af_cfg {
-    int force;  // Initialization type
     char **list; /* list of names of filters that are added to filter
                     list during first initialization of stream */
 };
@@ -107,9 +80,12 @@ struct af_stream {
     // The first and last filter in the list
     struct af_instance *first;
     struct af_instance *last;
-    // Storage for input and output data formats
+    // The user sets the input format (what the decoder outputs), and sets some
+    // or all fields in output to the output format the AO accepts.
+    // See fixup_output_format().
     struct mp_audio input;
     struct mp_audio output;
+    struct mp_audio filter_output;
     // Configuration for this stream
     struct af_cfg cfg;
     struct MPOpts *opts;
@@ -139,6 +115,9 @@ struct af_stream {
  * \param s filter chain
  */
 
+struct af_stream *af_new(struct MPOpts *opts);
+void af_destroy(struct af_stream *s);
+
 /**
  * \brief Initialize the stream "s".
  * \return 0 on success, -1 on failure
@@ -161,10 +140,10 @@ void af_uninit(struct af_stream *s);
 
 /**
  * \brief  Reinit the filter list from the given filter on downwards
- * \param  Filter instance to begin the reinit from
+ * See af.c.
  * \return AF_OK on success or AF_ERROR on failure
  */
-int af_reinit(struct af_stream *s, struct af_instance *af);
+int af_reinit(struct af_stream *s);
 
 /**
  * \brief This function adds the filter "name" to the stream s.
@@ -306,23 +285,13 @@ float af_softclip(float a);
 /** Print a list of all available audio filters */
 void af_help(void);
 
-/**
- * \brief fill the missing parameters in the struct mp_audio structure
- * \param data structure to fill
- * \ingroup af_filter
- *
- * Currently only sets bps based on format
- */
-void af_fix_parameters(struct mp_audio *data);
-
 /** Memory reallocation macro: if a local buffer is used (i.e. if the
    filter doesn't operate on the incoming buffer this macro must be
    called to ensure the buffer is big enough.
  * \ingroup af_filter
  */
 #define RESIZE_LOCAL_BUFFER(a, d) \
-    ((a->data->len < \
-      af_lencalc(a->mul, d)) ? af_resize_local_buffer(a, d) : AF_OK)
+    ((a->data->len < af_lencalc(a->mul, d)) ? af_resize_local_buffer(a, d) : AF_OK)
 
 /* Some other useful macro definitions*/
 #ifndef min
diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c
index aebcc3b201..f8003a70a3 100644
--- a/audio/filter/af_bs2b.c
+++ b/audio/filter/af_bs2b.c
@@ -105,9 +105,8 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
         format           = ((struct mp_audio*)arg)->format;
         af->data->rate   = ((struct mp_audio*)arg)->rate;
-        af->data->nch    = 2;     // bs2b is useful only for 2ch audio
-        af->data->bps    = ((struct mp_audio*)arg)->bps;
-        af->data->format = format;
+        mp_audio_set_num_channels(af->data, 2);     // bs2b is useful only for 2ch audio
+        mp_audio_set_format(af->data, format);
 
         /* check for formats supported by libbs2b
            and assign corresponding handlers */
@@ -162,8 +161,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
                 break;
             default:
                 af->play = play_f;
-                af->data->format = AF_FORMAT_FLOAT_NE;
-                af->data->bps = 4;
+                mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
                 break;
         }
 
diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c
index aa9aae8514..42571eea35 100644
--- a/audio/filter/af_center.c
+++ b/audio/filter/af_center.c
@@ -48,9 +48,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = max(s->ch+1,((struct mp_audio*)arg)->nch);
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
+    mp_audio_set_channels_old(af->data, max(s->ch+1,((struct mp_audio*)arg)->nch));
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     return af_test_output(af,(struct mp_audio*)arg);
   }
diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c
index 8f676d8cfd..7955018ec0 100644
--- a/audio/filter/af_channels.c
+++ b/audio/filter/af_channels.c
@@ -164,8 +164,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     }
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->format = ((struct mp_audio*)arg)->format;
-    af->data->bps    = ((struct mp_audio*)arg)->bps;
+    mp_audio_set_format(af->data, ((struct mp_audio*)arg)->format);
     af->mul          = (double)af->data->nch / ((struct mp_audio*)arg)->nch;
     return check_routes(s,((struct mp_audio*)arg)->nch,af->data->nch);
   case AF_CONTROL_COMMAND_LINE:{
@@ -194,54 +193,20 @@ static int control(struct af_instance* af, int cmd, void* arg)
       }
     }
 
-    if(AF_OK != af->control(af,AF_CONTROL_CHANNELS | AF_CONTROL_SET ,&nch))
+    struct mp_chmap chmap;
+    mp_chmap_from_channels(&chmap, nch);
+    if (AF_OK != af->control(af, AF_CONTROL_CHANNELS | AF_CONTROL_SET, &chmap))
       return AF_ERROR;
     return AF_OK;
   }
   case AF_CONTROL_CHANNELS | AF_CONTROL_SET:
     // Reinit must be called after this function has been called
 
-    // Sanity check
-    if(((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH){
-      mp_msg(MSGT_AFILTER, MSGL_ERR, "[channels] The number of output channels must be"
-	     " between 1 and %i. Current value is %i\n",AF_NCH,((int*)arg)[0]);
-      return AF_ERROR;
-    }
-
-    af->data->nch=((int*)arg)[0];
+    mp_audio_set_channels(af->data, (struct mp_chmap *)arg);
     if(!s->router)
       mp_msg(MSGT_AFILTER, MSGL_V, "[channels] Changing number of channels"
 	     " to %i\n",af->data->nch);
     return AF_OK;
-  case AF_CONTROL_CHANNELS | AF_CONTROL_GET:
-    *(int*)arg = af->data->nch;
-    return AF_OK;
-  case AF_CONTROL_CHANNELS_ROUTING | AF_CONTROL_SET:{
-    int ch = ((af_control_ext_t*)arg)->ch;
-    int* route = ((af_control_ext_t*)arg)->arg;
-    s->route[ch][FR] = route[FR];
-    s->route[ch][TO] = route[TO];
-    return AF_OK;
-  }
-  case AF_CONTROL_CHANNELS_ROUTING | AF_CONTROL_GET:{
-    int ch = ((af_control_ext_t*)arg)->ch;
-    int* route = ((af_control_ext_t*)arg)->arg;
-    route[FR] = s->route[ch][FR];
-    route[TO] = s->route[ch][TO];
-    return AF_OK;
-  }
-  case AF_CONTROL_CHANNELS_NR | AF_CONTROL_SET:
-    s->nr = *(int*)arg;
-    return AF_OK;
-  case AF_CONTROL_CHANNELS_NR | AF_CONTROL_GET:
-    *(int*)arg = s->nr;
-    return AF_OK;
-  case AF_CONTROL_CHANNELS_ROUTER | AF_CONTROL_SET:
-    s->router = *(int*)arg;
-    return AF_OK;
-  case AF_CONTROL_CHANNELS_ROUTER | AF_CONTROL_GET:
-    *(int*)arg = s->router;
-    return AF_OK;
   }
   return AF_UNKNOWN;
 }
@@ -277,7 +242,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
   // Set output data
   c->audio = l->audio;
   c->len   = c->len / c->nch * l->nch;
-  c->nch   = l->nch;
+  mp_audio_set_channels(c, &l->channels);
 
   return c;
 }
diff --git a/audio/filter/af_delay.c b/audio/filter/af_delay.c
index ce8d71980b..eb3e9e19cb 100644
--- a/audio/filter/af_delay.c
+++ b/audio/filter/af_delay.c
@@ -52,10 +52,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     for(i=0;i<af->data->nch;i++)
       free(s->q[i]);
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-    af->data->format = ((struct mp_audio*)arg)->format;
-    af->data->bps    = ((struct mp_audio*)arg)->bps;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
 
     // Allocate new delay queues
     for(i=0;i<af->data->nch;i++){
diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c
index ba1e10c0a7..e293f7f616 100644
--- a/audio/filter/af_drc.c
+++ b/audio/filter/af_drc.c
@@ -87,15 +87,10 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-
-    if(((struct mp_audio*)arg)->format == (AF_FORMAT_S16_NE)){
-      af->data->format = AF_FORMAT_S16_NE;
-      af->data->bps    = 2;
-    }else{
-      af->data->format = AF_FORMAT_FLOAT_NE;
-      af->data->bps    = 4;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+
+    if(((struct mp_audio*)arg)->format != (AF_FORMAT_S16_NE)){
+      mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
     }
     return af_test_output(af,(struct mp_audio*)arg);
   case AF_CONTROL_COMMAND_LINE:{
diff --git a/audio/filter/af_dummy.c b/audio/filter/af_dummy.c
index 29a5b3d4b8..5a54cdd80c 100644
--- a/audio/filter/af_dummy.c
+++ b/audio/filter/af_dummy.c
@@ -29,8 +29,8 @@
 static int control(struct af_instance* af, int cmd, void* arg)
 {
   switch(cmd){
-  case AF_CONTROL_REINIT:
-    memcpy(af->data,(struct mp_audio*)arg,sizeof(struct mp_audio));
+  case AF_CONTROL_REINIT: ;
+    *af->data = *(struct mp_audio*)arg;
     mp_msg(MSGT_AFILTER, MSGL_V, "[dummy] Was reinitialized: %iHz/%ich/%s\n",
 	af->data->rate,af->data->nch,af_fmt2str_short(af->data->format));
     return AF_OK;
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c
index c488ffaeaf..6441b9b116 100644
--- a/audio/filter/af_equalizer.c
+++ b/audio/filter/af_equalizer.c
@@ -96,10 +96,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     // Calculate number of active filters
     s->K=KM;
@@ -150,30 +148,6 @@ static int control(struct af_instance* af, int cmd, void* arg)
     }
     return AF_OK;
   }
-  case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_SET:{
-    float* gain = ((af_control_ext_t*)arg)->arg;
-    int    ch   = ((af_control_ext_t*)arg)->ch;
-    int    k;
-    if(ch >= AF_NCH || ch < 0)
-      return AF_ERROR;
-
-    for(k = 0 ; k<KM ; k++)
-      s->g[ch][k] = pow(10.0,clamp(gain[k],G_MIN,G_MAX)/20.0)-1.0;
-
-    return AF_OK;
-  }
-  case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_GET:{
-    float* gain = ((af_control_ext_t*)arg)->arg;
-    int    ch   = ((af_control_ext_t*)arg)->ch;
-    int    k;
-    if(ch >= AF_NCH || ch < 0)
-      return AF_ERROR;
-
-    for(k = 0 ; k<KM ; k++)
-      gain[k] = log10(s->g[ch][k]+1.0) * 20.0;
-
-    return AF_OK;
-  }
   }
   return AF_UNKNOWN;
 }
diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c
index 5e3a1869ee..ea0aa938ca 100644
--- a/audio/filter/af_export.c
+++ b/audio/filter/af_export.c
@@ -85,10 +85,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
       close(s->fd);
 
     // Accept only int16_t as input format (which sucks)
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-    af->data->format = AF_FORMAT_S16_NE;
-    af->data->bps    = 2;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 
     // If buffer length isn't set, set it to the default value
     if(s->sz == 0)
diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c
index 0f7fe36861..ed1fd27898 100644
--- a/audio/filter/af_extrastereo.c
+++ b/audio/filter/af_extrastereo.c
@@ -47,17 +47,14 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = 2;
-    if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT_NE)
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_num_channels(af->data, 2);
+    if (af->data->format == AF_FORMAT_FLOAT_NE)
     {
-	af->data->format = AF_FORMAT_FLOAT_NE;
-	af->data->bps = 4;
 	af->play = play_float;
     }// else
     {
-	af->data->format = AF_FORMAT_S16_NE;
-	af->data->bps = 2;
+        mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 	af->play = play_s16;
     }
 
@@ -69,12 +66,6 @@ static int control(struct af_instance* af, int cmd, void* arg)
     s->mul = f;
     return AF_OK;
   }
-  case AF_CONTROL_ES_MUL | AF_CONTROL_SET:
-    s->mul = *(float*)arg;
-    return AF_OK;
-  case AF_CONTROL_ES_MUL | AF_CONTROL_GET:
-    *(float*)arg = s->mul;
-    return AF_OK;
   }
   return AF_UNKNOWN;
 }
diff --git a/audio/filter/af_force.c b/audio/filter/af_force.c
new file mode 100644
index 0000000000..51fe83d0f0
--- /dev/null
+++ b/audio/filter/af_force.c
@@ -0,0 +1,146 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdlib.h>
+
+#include <libavutil/common.h>
+
+#include "core/m_config.h"
+#include "core/m_option.h"
+
+#include "audio/format.h"
+#include "af.h"
+
+struct priv {
+    struct m_config *config;
+
+    int in_format;
+    int in_srate;
+    struct mp_chmap in_channels;
+    int out_format;
+    int out_srate;
+    struct mp_chmap out_channels;
+
+    struct mp_audio data;
+    struct mp_audio temp;
+};
+
+static const struct priv defaults = {
+    .in_format = AF_FORMAT_UNKNOWN,
+    .out_format = AF_FORMAT_UNKNOWN,
+};
+
+#define OPT_BASE_STRUCT struct priv
+
+static const struct m_option options[] = {
+    OPT_AUDIOFORMAT("format", in_format, 0),
+    OPT_INTRANGE("srate", in_srate, 0, 1000, 8*48000),
+    OPT_CHMAP("channels", in_channels, CONF_MIN, .min = 0),
+    OPT_AUDIOFORMAT("out-format", out_format, 0),
+    OPT_INTRANGE("out-srate", out_srate, 0, 1000, 8*48000),
+    OPT_CHMAP("out-channels", out_channels, CONF_MIN, .min = 0),
+    {0}
+};
+
+static int control(struct af_instance *af, int cmd, void *arg)
+{
+    struct priv *priv = af->setup;
+
+    switch (cmd) {
+    case AF_CONTROL_REINIT: {
+        struct mp_audio *in = arg;
+        struct mp_audio orig_in = *in;
+        struct mp_audio *out = af->data;
+
+        if (priv->in_format != AF_FORMAT_UNKNOWN)
+            mp_audio_set_format(in, priv->in_format);
+
+        if (priv->in_channels.num)
+            mp_audio_set_channels(in, &priv->in_channels);
+
+        if (priv->in_srate)
+            in->rate = priv->in_srate;
+
+        mp_audio_copy_config(out, in);
+
+        if (priv->out_format != AF_FORMAT_UNKNOWN)
+            mp_audio_set_format(out, priv->out_format);
+
+        if (priv->out_channels.num)
+            mp_audio_set_channels(out, &priv->out_channels);
+
+        if (priv->out_srate)
+            out->rate = priv->out_srate;
+
+        if (in->nch != out->nch || in->bps != out->bps) {
+            mp_msg(MSGT_AFILTER, MSGL_ERR,
+                   "[af_force] Forced input/output format are incompatible.\n");
+            return AF_ERROR;
+        }
+
+        return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
+    }
+    case AF_CONTROL_COMMAND_LINE: {
+        if (m_config_parse_suboptions(priv->config, "af_force", (char *)arg) < 0)
+            return AF_ERROR;
+        return AF_OK;
+    }
+    }
+    return AF_UNKNOWN;
+}
+
+static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
+{
+    struct priv *priv = af->setup;
+    struct mp_audio *r = &priv->temp;
+
+    *r = *af->data;
+    r->audio = data->audio;
+    r->len = data->len;
+
+    return r;
+}
+
+static void uninit(struct af_instance *af)
+{
+    talloc_free(af->setup);
+}
+
+static int af_open(struct af_instance *af)
+{
+    af->control = control;
+    af->uninit = uninit;
+    af->play = play;
+    af->mul = 1;
+    struct priv *priv = talloc(NULL, struct priv);
+    af->setup = priv;
+    *priv = defaults;
+    priv->config = m_config_simple(priv);
+    talloc_steal(priv, priv->config);
+    m_config_register_options(priv->config, options);
+    af->data = &priv->data;
+    return AF_OK;
+}
+
+struct af_info af_info_force = {
+    "Force audio format",
+    "force",
+    "",
+    "",
+    0,
+    af_open
+};
diff --git a/audio/filter/af_format.c b/audio/filter/af_format.c
index 37d29c1f80..616bc9e494 100644
--- a/audio/filter/af_format.c
+++ b/audio/filter/af_format.c
@@ -75,6 +75,14 @@ static int check_format(int format)
   return AF_ERROR;
 }
 
+static bool test_conversion(int src_format, int dst_format)
+{
+    // This is the fallback conversion filter, so this filter is always
+    // inserted on format mismatches if no other filter can handle it.
+    // Initializing the filter might still fail.
+    return true;
+}
+
 // Initialization and runtime control
 static int control(struct af_instance* af, int cmd, void* arg)
 {
@@ -86,8 +94,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     int supported_ac3 = 0;
 
     // Make sure this filter isn't redundant
-    if(af->data->format == data->format &&
-       af->data->bps == data->bps)
+    if(af->data->format == data->format)
       return AF_DETACH;
 
     // A bit complex because we can convert AC3
@@ -113,7 +120,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 	   buf1, buf2);
 
     af->data->rate = data->rate;
-    af->data->nch  = data->nch;
+    mp_audio_set_channels(af->data, &data->channels);
     af->mul        = (double)af->data->bps / data->bps;
 
     af->play = play; // set default
@@ -147,7 +154,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
       mp_msg(MSGT_AFILTER, MSGL_ERR, "[format] %s is not a valid format\n", (char *)arg);
       return AF_ERROR;
     }
-    if(AF_OK != af->control(af,AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET,&format))
+    if(AF_OK != af->control(af, AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET,&format))
       return AF_ERROR;
     return AF_OK;
   }
@@ -156,8 +163,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!AF_FORMAT_IS_AC3(*(int*)arg) && AF_OK != check_format(*(int*)arg))
       return AF_ERROR;
 
-    af->data->format = *(int*)arg;
-    af->data->bps = af_fmt2bits(af->data->format)/8;
+    mp_audio_set_format(af->data, *(int*)arg);
 
     return AF_OK;
   }
@@ -186,7 +192,7 @@ static struct mp_audio* play_swapendian(struct af_instance* af, struct mp_audio*
   endian(c->audio,l->audio,len,c->bps);
 
   c->audio = l->audio;
-  c->format = l->format;
+  mp_audio_set_format(c, l->format);
 
   return c;
 }
@@ -203,9 +209,8 @@ static struct mp_audio* play_float_s16(struct af_instance* af, struct mp_audio*
   float2int(c->audio, l->audio, len, 2);
 
   c->audio = l->audio;
+  mp_audio_set_format(c, l->format);
   c->len = len*2;
-  c->bps = 2;
-  c->format = l->format;
 
   return c;
 }
@@ -222,9 +227,8 @@ static struct mp_audio* play_s16_float(struct af_instance* af, struct mp_audio*
   int2float(c->audio, l->audio, len, 2);
 
   c->audio = l->audio;
+  mp_audio_set_format(c, l->format);
   c->len = len*4;
-  c->bps = 4;
-  c->format = l->format;
 
   return c;
 }
@@ -276,9 +280,8 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
 
   // Set output data
   c->audio  = l->audio;
+  mp_audio_set_format(c, l->format);
   c->len    = len*l->bps;
-  c->bps    = l->bps;
-  c->format = l->format;
   return c;
 }
 
@@ -301,7 +304,8 @@ struct af_info af_info_format = {
   "Anders",
   "",
   AF_FLAGS_REENTRANT,
-  af_open
+  af_open,
+  .test_conversion = test_conversion,
 };
 
 static inline uint32_t load24bit(void* data, int pos) {
diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
index 4f5eedb29d..85e6477b31 100644
--- a/audio/filter/af_hrtf.c
+++ b/audio/filter/af_hrtf.c
@@ -299,7 +299,7 @@ static int control(struct af_instance *af, int cmd, void* arg)
 		   af->data->rate);
 	    return AF_ERROR;
 	}
-	af->data->nch    = ((struct mp_audio*)arg)->nch;
+	mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch);
 	    if(af->data->nch == 2) {
  	       /* 2 channel input */
  	       if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
@@ -308,13 +308,12 @@ static int control(struct af_instance *af, int cmd, void* arg)
 	       }
 	    }
 	    else if (af->data->nch < 5)
-	      af->data->nch = 5;
-	af->data->format = AF_FORMAT_S16_NE;
-	af->data->bps    = 2;
+	      mp_audio_set_channels_old(af->data, 5);
+        mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 	test_output_res = af_test_output(af, (struct mp_audio*)arg);
 	af->mul = 2.0 / af->data->nch;
 	// after testing input set the real output format
-	af->data->nch = 2;
+        mp_audio_set_num_channels(af->data, 2);
 	s->print_flag = 1;
 	return test_output_res;
     case AF_CONTROL_COMMAND_LINE:
@@ -566,7 +565,7 @@ static struct mp_audio* play(struct af_instance *af, struct mp_audio *data)
     /* Set output data */
     data->audio = af->data->audio;
     data->len   = data->len / data->nch * 2;
-    data->nch   = 2;
+    mp_audio_set_num_channels(data, 2);
 
     return data;
 }
diff --git a/audio/filter/af_karaoke.c b/audio/filter/af_karaoke.c
index 965eb8f40d..faed389625 100644
--- a/audio/filter/af_karaoke.c
+++ b/audio/filter/af_karaoke.c
@@ -34,10 +34,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
 {
 	switch(cmd){
 		case AF_CONTROL_REINIT:
-		af->data->rate	= ((struct mp_audio*)arg)->rate;
-		af->data->nch	= ((struct mp_audio*)arg)->nch;
-		af->data->format= AF_FORMAT_FLOAT_NE;
-		af->data->bps	= 4;
+                mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+                mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 		return af_test_output(af,(struct mp_audio*)arg);
 	}
 	return AF_UNKNOWN;
diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c
index c1b3f24360..5e516f060e 100644
--- a/audio/filter/af_ladspa.c
+++ b/audio/filter/af_ladspa.c
@@ -291,7 +291,7 @@ static int af_ladspa_parse_plugin(af_ladspa_t *setup) {
 static void* mydlopen(const char *filename, int flag) {
     char *buf;
     const char *end, *start, *ladspapath;
-    int endsinso, needslash;
+    int endsinso;
     size_t filenamelen;
     void *result = NULL;
 
@@ -324,9 +324,9 @@ static void* mydlopen(const char *filename, int flag) {
     ladspapath=getenv("LADSPA_PATH");
 
     if (ladspapath) {
-
         start=ladspapath;
         while (*start != '\0') {
+            int needslash;
             end=start;
             while ( (*end != ':') && (*end != '\0') )
                 end++;
@@ -487,7 +487,6 @@ static int af_ladspa_malloc_failed(char *myname) {
 
 static int control(struct af_instance *af, int cmd, void *arg) {
     af_ladspa_t *setup = (af_ladspa_t*) af->setup;
-    int i, r;
     float val;
 
     switch(cmd) {
@@ -498,10 +497,8 @@ static int control(struct af_instance *af, int cmd, void *arg) {
 
         /* accept FLOAT, let af_format do conversion */
 
-        af->data->rate   = ((struct mp_audio*)arg)->rate;
-        af->data->nch    = ((struct mp_audio*)arg)->nch;
-        af->data->format = AF_FORMAT_FLOAT_NE;
-        af->data->bps    = 4;
+        mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+        mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
         /* arg->len is not set here yet, so init of buffers and connecting the
          * filter, has to be done in play() :-/
@@ -538,7 +535,10 @@ static int control(struct af_instance *af, int cmd, void *arg) {
         }
         line += strlen(buf);
         setup->file = strdup(buf);
-        if (!setup->file) return af_ladspa_malloc_failed(setup->myname);
+        if (!setup->file) {
+            free(buf);
+            return af_ladspa_malloc_failed(setup->myname);
+        }
         mp_msg(MSGT_AFILTER, MSGL_V, "%s: file --> %s\n", setup->myname,
                                                         setup->file);
         if (*line != '\0') line++; /* read ':' */
@@ -554,7 +554,10 @@ static int control(struct af_instance *af, int cmd, void *arg) {
         }
         line += strlen(buf);
         setup->label = strdup(buf);
-        if (!setup->label) return af_ladspa_malloc_failed(setup->myname);
+        if (!setup->label) {
+            free(buf);
+            return af_ladspa_malloc_failed(setup->myname);
+        }
         mp_msg(MSGT_AFILTER, MSGL_V, "%s: label --> %s\n", setup->myname,
                                                                 setup->label);
 /*        if (*line != '0') line++; */ /* read ':' */
@@ -581,15 +584,14 @@ static int control(struct af_instance *af, int cmd, void *arg) {
 
         /* ninputcontrols is set by now, read control values from arg */
 
-        for(i=0; i<setup->ninputcontrols; i++) {
+        for (int i = 0; i < setup->ninputcontrols; i++) {
             if (!line || *line != ':') {
                 mp_msg(MSGT_AFILTER, MSGL_ERR, "%s: %s\n", setup->myname,
                                         _("Not enough controls specified on the command line."));
                 return AF_ERROR;
             }
             line++;
-            r = sscanf(line, "%f", &val);
-            if (r!=1) {
+            if (sscanf(line, "%f", &val) != 1) {
                 mp_msg(MSGT_AFILTER, MSGL_ERR, "%s: %s\n", setup->myname,
                                         _("Not enough controls specified on the command line."));
                 return AF_ERROR;
@@ -599,7 +601,7 @@ static int control(struct af_instance *af, int cmd, void *arg) {
         }
 
         mp_msg(MSGT_AFILTER, MSGL_V, "%s: input controls: ", setup->myname);
-        for(i=0; i<setup->ninputcontrols; i++) {
+        for (int i = 0; i < setup->ninputcontrols; i++) {
             mp_msg(MSGT_AFILTER, MSGL_V, "%0.4f ",
                             setup->inputcontrols[setup->inputcontrolsmap[i]]);
         }
@@ -609,7 +611,7 @@ static int control(struct af_instance *af, int cmd, void *arg) {
 
         mp_msg(MSGT_AFILTER, MSGL_V, "%s: checking boundaries of input controls\n",
                                                                 setup->myname);
-        for(i=0; i<setup->ninputcontrols; i++) {
+        for (int i = 0; i < setup->ninputcontrols; i++) {
             int p = setup->inputcontrolsmap[i];
             LADSPA_PortRangeHint hint =
                                 setup->plugin_descriptor->PortRangeHints[p];
@@ -651,8 +653,6 @@ static int control(struct af_instance *af, int cmd, void *arg) {
  */
 
 static void uninit(struct af_instance *af) {
-    int i;
-
     free(af->data);
     if (af->setup) {
         af_ladspa_t *setup = (af_ladspa_t*) af->setup;
@@ -664,7 +664,7 @@ static void uninit(struct af_instance *af) {
         }
 
         if (setup->chhandles) {
-            for(i=0; i<setup->nch; i+=setup->ninputs) {
+            for (int i = 0; i < setup->nch; i+=setup->ninputs) {
                 if (pdes->deactivate) pdes->deactivate(setup->chhandles[i]);
                 if (pdes->cleanup) pdes->cleanup(setup->chhandles[i]);
             }
@@ -681,13 +681,13 @@ static void uninit(struct af_instance *af) {
         free(setup->outputs);
 
         if (setup->inbufs) {
-            for(i=0; i<setup->nch; i++)
+            for(int i = 0; i < setup->nch; i++)
                 free(setup->inbufs[i]);
             free(setup->inbufs);
         }
 
         if (setup->outbufs) {
-            for(i=0; i<setup->nch; i++)
+            for (int i = 0; i < setup->nch; i++)
                 free(setup->outbufs[i]);
             free(setup->outbufs);
         }
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index 94d59a11ca..7eacc01d81 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -75,16 +75,16 @@ static int control(struct af_instance *af, int cmd, void *arg)
         if (AF_FORMAT_IS_AC3(data->format) || data->nch < s->min_channel_num)
             return AF_DETACH;
 
-        af->data->format = s->in_sampleformat;
-        af->data->bps = af_fmt2bits(s->in_sampleformat) / 8;
+        mp_audio_set_format(af->data, s->in_sampleformat);
         if (data->rate == 48000 || data->rate == 44100 || data->rate == 32000)
             af->data->rate = data->rate;
         else
             af->data->rate = 48000;
         if (data->nch > AC3_MAX_CHANNELS)
-            af->data->nch = AC3_MAX_CHANNELS;
+            mp_audio_set_num_channels(af->data, AC3_MAX_CHANNELS);
         else
-            af->data->nch = data->nch;
+            mp_audio_set_channels(af->data, &data->channels);
+        mp_chmap_reorder_to_lavc(&af->data->channels);
         test_output_res = af_test_output(af, data);
 
         s->pending_len = 0;
@@ -108,8 +108,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
             // Put sample parameters
             s->lavc_actx->channels = af->data->nch;
-            s->lavc_actx->channel_layout =
-                av_get_default_channel_layout(af->data->nch);
+            s->lavc_actx->channel_layout = mp_chmap_to_lavc(&af->data->channels);
             s->lavc_actx->sample_rate = af->data->rate;
             s->lavc_actx->bit_rate = bit_rate;
 
@@ -123,9 +122,8 @@ static int control(struct af_instance *af, int cmd, void *arg)
                    "encoder frame size %d\n", s->lavc_actx->frame_size);
             return AF_ERROR;
         }
-        af->data->format = AF_FORMAT_AC3_BE;
-        af->data->bps = 2;
-        af->data->nch = 2;
+        mp_audio_set_format(af->data, AF_FORMAT_AC3_BE);
+        mp_audio_set_num_channels(af->data, 2);
         return test_output_res;
     case AF_CONTROL_COMMAND_LINE:
         mp_msg(MSGT_AFILTER, MSGL_DBG2, "af_lavcac3enc cmdline: %s.\n", (char*)arg);
@@ -235,15 +233,6 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
             src2= s->pending_data;
         }
 
-        if (c->nch >= 5) {
-            reorder_channel_nch(src2,
-                                AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
-                                AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
-                                c->nch,
-                                s->expect_len / samplesize,
-                                samplesize);
-        }
-
         void *data = (void *) src2;
         if (s->planarize) {
             void *data2 = malloc(s->expect_len);
@@ -316,8 +305,8 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
         buf += len;
     }
     c->audio = l->audio;
-    c->nch   = 2;
-    c->bps   = 2;
+    mp_audio_set_num_channels(c, 2);
+    mp_audio_set_format(c, af->data->format);
     c->len   = outsize;
     mp_msg(MSGT_AFILTER, MSGL_DBG2, "play return size %d, pending %d\n",
            outsize, s->pending_len);
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
index 5b26a0dce6..4372100ea0 100644
--- a/audio/filter/af_lavrresample.c
+++ b/audio/filter/af_lavrresample.c
@@ -43,6 +43,7 @@
 #define avresample_available(x) 0
 #define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \
     swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples)
+#define avresample_set_channel_mapping swr_set_channel_mapping
 #else
 #error "config.h broken"
 #endif
@@ -50,6 +51,7 @@
 #include "core/mp_msg.h"
 #include "core/subopt-helper.h"
 #include "audio/filter/af.h"
+#include "audio/fmt-conversion.h"
 
 struct af_resample_opts {
     int filter_size;
@@ -57,14 +59,24 @@ struct af_resample_opts {
     int linear;
     double cutoff;
 
-    int out_rate;
     int in_rate;
+    int in_format;
+    struct mp_chmap in_channels;
+    int out_rate;
+    int out_format;
+    struct mp_chmap out_channels;
 };
 
 struct af_resample {
+    int allow_detach;
     struct AVAudioResampleContext *avrctx;
+    struct AVAudioResampleContext *avrctx_out; // for output channel reordering
     struct af_resample_opts ctx;   // opts in the context
     struct af_resample_opts opts;  // opts requested by the user
+    // At least libswresample keeps a pointer around for this:
+    int reorder_in[MP_NUM_CHANNELS];
+    int reorder_out[MP_NUM_CHANNELS];
+    uint8_t *reorder_buffer;
 };
 
 #ifdef CONFIG_LIBAVRESAMPLE
@@ -88,8 +100,12 @@ static bool needs_lavrctx_reconfigure(struct af_resample *s,
                                       struct mp_audio *in,
                                       struct mp_audio *out)
 {
-    return s->ctx.out_rate    != out->rate ||
-           s->ctx.in_rate     != in->rate ||
+    return s->ctx.in_rate     != in->rate ||
+           s->ctx.in_format   != in->format ||
+           !mp_chmap_equals(&s->ctx.in_channels, &in->channels) ||
+           s->ctx.out_rate    != out->rate ||
+           s->ctx.out_format  != out->format ||
+           !mp_chmap_equals(&s->ctx.out_channels, &out->channels) ||
            s->ctx.filter_size != s->opts.filter_size ||
            s->ctx.phase_shift != s->opts.phase_shift ||
            s->ctx.linear      != s->opts.linear ||
@@ -97,6 +113,12 @@ static bool needs_lavrctx_reconfigure(struct af_resample *s,
 
 }
 
+static bool test_conversion(int src_format, int dst_format)
+{
+    return af_to_avformat(src_format) != AV_SAMPLE_FMT_NONE &&
+           af_to_avformat(dst_format) != AV_SAMPLE_FMT_NONE;
+}
+
 #define ctx_opt_set_int(a,b) av_opt_set_int(s->avrctx, (a), (b), 0)
 #define ctx_opt_set_dbl(a,b) av_opt_set_double(s->avrctx, (a), (b), 0)
 
@@ -108,36 +130,70 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
     switch (cmd) {
     case AF_CONTROL_REINIT: {
-        if ((out->rate == in->rate) || (out->rate == 0))
+        struct mp_audio orig_in = *in;
+
+        if (((out->rate    == in->rate) || (out->rate == 0)) &&
+            (out->format   == in->format) &&
+            (mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
+            s->allow_detach)
             return AF_DETACH;
 
-        out->nch    = FFMIN(in->nch, AF_NCH);
-        out->format = AF_FORMAT_S16_NE;
-        out->bps    = 2;
-        af->mul     = (double) out->rate / in->rate;
+        if (out->rate == 0)
+            out->rate = in->rate;
+
+        if (mp_chmap_is_empty(&out->channels))
+            mp_audio_set_channels(out, &in->channels);
+
+        enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
+        if (in_samplefmt == AV_SAMPLE_FMT_NONE) {
+            mp_audio_set_format(in, AF_FORMAT_FLOAT_NE);
+            in_samplefmt = af_to_avformat(in->format);
+        }
+        enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
+        if (out_samplefmt == AV_SAMPLE_FMT_NONE) {
+            mp_audio_set_format(out, in->format);
+            out_samplefmt = in_samplefmt;
+        }
+
+        af->mul     = (double) (out->rate * out->nch) / (in->rate * in->nch);
         af->delay   = out->nch * s->opts.filter_size / FFMIN(af->mul, 1);
 
         if (needs_lavrctx_reconfigure(s, in, out)) {
-            if (s->avrctx)
-                avresample_close(s->avrctx);
+            avresample_close(s->avrctx);
+            avresample_close(s->avrctx_out);
 
             s->ctx.out_rate    = out->rate;
             s->ctx.in_rate     = in->rate;
+            s->ctx.out_format  = out->format;
+            s->ctx.in_format   = in->format;
+            s->ctx.out_channels= out->channels;
+            s->ctx.in_channels = in->channels;
             s->ctx.filter_size = s->opts.filter_size;
             s->ctx.phase_shift = s->opts.phase_shift;
             s->ctx.linear      = s->opts.linear;
             s->ctx.cutoff      = s->opts.cutoff;
 
-            int ch_layout = av_get_default_channel_layout(out->nch);
+            struct mp_chmap map_in = in->channels;
+            struct mp_chmap map_out = out->channels;
+
+            // Try not to do any remixing if at least one is "unknown".
+            if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&map_out)) {
+                mp_chmap_set_unknown(&map_in, map_in.num);
+                mp_chmap_set_unknown(&map_out, map_out.num);
+            }
+
+            // unchecked: don't take any channel reordering into account
+            uint64_t in_ch_layout = mp_chmap_to_lavc_unchecked(&map_in);
+            uint64_t out_ch_layout = mp_chmap_to_lavc_unchecked(&map_out);
 
-            ctx_opt_set_int("in_channel_layout",  ch_layout);
-            ctx_opt_set_int("out_channel_layout", ch_layout);
+            ctx_opt_set_int("in_channel_layout",  in_ch_layout);
+            ctx_opt_set_int("out_channel_layout", out_ch_layout);
 
             ctx_opt_set_int("in_sample_rate",     s->ctx.in_rate);
             ctx_opt_set_int("out_sample_rate",    s->ctx.out_rate);
 
-            ctx_opt_set_int("in_sample_fmt",      AV_SAMPLE_FMT_S16);
-            ctx_opt_set_int("out_sample_fmt",     AV_SAMPLE_FMT_S16);
+            ctx_opt_set_int("in_sample_fmt",      in_samplefmt);
+            ctx_opt_set_int("out_sample_fmt",     out_samplefmt);
 
             ctx_opt_set_int("filter_size",        s->ctx.filter_size);
             ctx_opt_set_int("phase_shift",        s->ctx.phase_shift);
@@ -145,20 +201,51 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
             ctx_opt_set_dbl("cutoff",             s->ctx.cutoff);
 
-            if (avresample_open(s->avrctx) < 0) {
+            struct mp_chmap in_lavc;
+            mp_chmap_from_lavc(&in_lavc, in_ch_layout);
+            mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
+
+            struct mp_chmap out_lavc;
+            mp_chmap_from_lavc(&out_lavc, out_ch_layout);
+            mp_chmap_get_reorder(s->reorder_out, &out_lavc, &map_out);
+
+            // Same configuration; we just reorder.
+            av_opt_set_int(s->avrctx_out, "in_channel_layout", out_ch_layout, 0);
+            av_opt_set_int(s->avrctx_out, "out_channel_layout", out_ch_layout, 0);
+            av_opt_set_int(s->avrctx_out, "in_sample_fmt", out_samplefmt, 0);
+            av_opt_set_int(s->avrctx_out, "out_sample_fmt", out_samplefmt, 0);
+            av_opt_set_int(s->avrctx_out, "in_sample_rate", s->ctx.out_rate, 0);
+            av_opt_set_int(s->avrctx_out, "out_sample_rate", s->ctx.out_rate, 0);
+
+            // API has weird requirements, quoting avresample.h:
+            //  * This function can only be called when the allocated context is not open.
+            //  * Also, the input channel layout must have already been set.
+            avresample_set_channel_mapping(s->avrctx, s->reorder_in);
+            avresample_set_channel_mapping(s->avrctx_out, s->reorder_out);
+
+            if (avresample_open(s->avrctx) < 0 ||
+                avresample_open(s->avrctx_out) < 0)
+            {
                 mp_msg(MSGT_AFILTER, MSGL_ERR, "[lavrresample] Cannot open "
                        "Libavresample Context. \n");
                 return AF_ERROR;
             }
         }
 
-        int out_rate, test_output_res;
-        // hack to make af_test_output ignore the samplerate change
-        out_rate  = out->rate;
-        out->rate = in->rate;
-        test_output_res = af_test_output(af, in);
-        out->rate = out_rate;
-        return test_output_res;
+        return ((in->format == orig_in.format) &&
+                mp_chmap_equals(&in->channels, &orig_in.channels))
+               ? AF_OK : AF_FALSE;
+    }
+    case AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET: {
+        if (af_to_avformat(*(int*)arg) == AV_SAMPLE_FMT_NONE)
+            return AF_FALSE;
+
+        mp_audio_set_format(af->data, *(int*)arg);
+        return AF_OK;
+    }
+    case AF_CONTROL_CHANNELS | AF_CONTROL_SET: {
+        mp_audio_set_channels(af->data, (struct mp_chmap *)arg);
+        return AF_OK;
     }
     case AF_CONTROL_COMMAND_LINE: {
         s->opts.cutoff = 0.0;
@@ -169,6 +256,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
             {"phase_shift",  OPT_ARG_INT,    &s->opts.phase_shift, NULL},
             {"linear",       OPT_ARG_BOOL,   &s->opts.linear, NULL},
             {"cutoff",       OPT_ARG_FLOAT,  &s->opts.cutoff, NULL},
+            {"detach",       OPT_ARG_BOOL,   &s->allow_detach, NULL},
             {0}
         };
 
@@ -198,10 +286,21 @@ static void uninit(struct af_instance *af)
         struct af_resample *s = af->setup;
         if (s->avrctx)
             avresample_close(s->avrctx);
+        if (s->avrctx_out)
+            avresample_close(s->avrctx_out);
         talloc_free(af->setup);
     }
 }
 
+static bool needs_reorder(int *reorder, int num_ch)
+{
+    for (int n = 0; n < num_ch; n++) {
+        if (reorder[n] != n)
+            return true;
+    }
+    return false;
+}
+
 static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
 {
     struct af_resample *s = af->setup;
@@ -227,10 +326,18 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
             (uint8_t **) &out->audio, out_size, out_samples,
             (uint8_t **) &in->audio,  in_size,  in_samples);
 
-    out_size  = out->bps * out_samples * out->nch;
-    in->audio = out->audio;
-    in->len   = out_size;
-    in->rate  = s->ctx.out_rate;
+    *data = *out;
+
+    if (needs_reorder(s->reorder_out, out->nch)) {
+        if (talloc_get_size(s->reorder_buffer) < out_size)
+            s->reorder_buffer = talloc_realloc_size(s, s->reorder_buffer, out_size);
+        data->audio = s->reorder_buffer;
+        out_samples = avresample_convert(s->avrctx_out,
+                (uint8_t **) &data->audio, out_size, out_samples,
+                (uint8_t **) &out->audio, out_size, out_samples);
+    }
+
+    data->len = out->bps * out_samples * out->nch;
     return data;
 }
 
@@ -244,7 +351,7 @@ static int af_open(struct af_instance *af)
     af->mul     = 1;
     af->data    = talloc_zero(s, struct mp_audio);
 
-    af->data->rate   = 44100;
+    af->data->rate   = 0;
 
     int default_filter_size = 16;
     s->opts = (struct af_resample_opts) {
@@ -254,10 +361,13 @@ static int af_open(struct af_instance *af)
         .phase_shift = 10,
     };
 
+    s->allow_detach = 1;
+
     s->avrctx = avresample_alloc_context();
+    s->avrctx_out = avresample_alloc_context();
     af->setup = s;
 
-    if (s->avrctx) {
+    if (s->avrctx && s->avrctx_out) {
         return AF_OK;
     } else {
         mp_msg(MSGT_AFILTER, MSGL_ERR, "[lavrresample] Cannot initialize "
@@ -273,5 +383,6 @@ struct af_info af_info_lavrresample = {
     "Stefano Pigozzi (based on Michael Niedermayer's lavcresample)",
     "",
     AF_FLAGS_REENTRANT,
-    af_open
+    af_open,
+    .test_conversion = test_conversion,
 };
diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c
index 8b1783ee84..d6f7538868 100644
--- a/audio/filter/af_pan.c
+++ b/audio/filter/af_pan.c
@@ -34,6 +34,15 @@ typedef struct af_pan_s
   float level[AF_NCH][AF_NCH];	// Gain level for each channel
 }af_pan_t;
 
+static void set_channels(struct mp_audio *mpa, int num)
+{
+    struct mp_chmap map;
+    // "unknown" channel layouts make it easier to pass through audio data,
+    // without triggering remixing.
+    mp_chmap_set_unknown(&map, num);
+    mp_audio_set_channels(mpa, &map);
+}
+
 // Initialization and runtime control
 static int control(struct af_instance* af, int cmd, void* arg)
 {
@@ -45,15 +54,13 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
-    af->data->nch    = s->nch ? s->nch: ((struct mp_audio*)arg)->nch;
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+    set_channels(af->data, s->nch ? s->nch: ((struct mp_audio*)arg)->nch);
     af->mul          = (double)af->data->nch / ((struct mp_audio*)arg)->nch;
 
     if((af->data->format != ((struct mp_audio*)arg)->format) ||
        (af->data->bps != ((struct mp_audio*)arg)->bps)){
-      ((struct mp_audio*)arg)->format = af->data->format;
-      ((struct mp_audio*)arg)->bps = af->data->bps;
+      mp_audio_set_format((struct mp_audio*)arg, af->data->format);
       return AF_FALSE;
     }
     return AF_OK;
@@ -109,14 +116,11 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH){
       mp_msg(MSGT_AFILTER, MSGL_ERR, "[pan] The number of output channels must be"
-	     " between 1 and %i. Current value is %i\n",AF_NCH,((int*)arg)[0]);
+            " between 1 and %i. Current value is %i\n",AF_NCH,((int*)arg)[0]);
       return AF_ERROR;
     }
     s->nch=((int*)arg)[0];
     return AF_OK;
-  case AF_CONTROL_PAN_NOUT | AF_CONTROL_GET:
-    *(int*)arg = af->data->nch;
-    return AF_OK;
   case AF_CONTROL_PAN_BALANCE | AF_CONTROL_SET:{
     float val = *(float*)arg;
     if (s->nch)
@@ -181,7 +185,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
   // Set output data
   c->audio = l->audio;
   c->len   = c->len / c->nch * l->nch;
-  c->nch   = l->nch;
+  set_channels(c, l->nch);
 
   return c;
 }
diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c
index cf326fedfb..5cf0f3b082 100644
--- a/audio/filter/af_scaletempo.c
+++ b/audio/filter/af_scaletempo.c
@@ -302,26 +302,22 @@ static int control(struct af_instance* af, int cmd, void* arg)
            "[scaletempo] %.3f speed * %.3f scale_nominal = %.3f\n",
            s->speed, s->scale_nominal, s->scale);
 
+    mp_audio_copy_config(af->data, data);
+
     if (s->scale == 1.0) {
       if (s->speed_tempo && s->speed_pitch)
         return AF_DETACH;
-      memcpy(af->data, data, sizeof(struct mp_audio));
       af->delay = 0;
       af->mul = 1;
       return af_test_output(af, data);
     }
 
-    af->data->rate = data->rate;
-    af->data->nch  = data->nch;
-    if ( data->format == AF_FORMAT_S16_LE
-         || data->format == AF_FORMAT_S16_BE ) {
+    if (data->format == AF_FORMAT_S16_NE) {
       use_int = 1;
-      af->data->format  = AF_FORMAT_S16_NE;
-      af->data->bps     = bps = 2;
     } else {
-      af->data->format = AF_FORMAT_FLOAT_NE;
-      af->data->bps    = bps = 4;
+      mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
     }
+    bps = af->data->bps;
 
     frames_stride           = srate * s->ms_stride;
     s->bytes_stride         = frames_stride * bps * nch;
diff --git a/audio/filter/af_sinesuppress.c b/audio/filter/af_sinesuppress.c
index 36f7189f00..10f0b650ec 100644
--- a/audio/filter/af_sinesuppress.c
+++ b/audio/filter/af_sinesuppress.c
@@ -54,8 +54,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = 1;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_num_channels(af->data, 1);
 #if 0
     if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT_NE)
     {
@@ -65,8 +65,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     }// else
 #endif
     {
-	af->data->format = AF_FORMAT_S16_NE;
-	af->data->bps = 2;
+        mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 	af->play = play_s16;
     }
 
@@ -79,18 +78,6 @@ static int control(struct af_instance* af, int cmd, void* arg)
     s->decay = f2;
     return AF_OK;
   }
-  case AF_CONTROL_SS_FREQ | AF_CONTROL_SET:
-    s->freq = *(float*)arg;
-    return AF_OK;
-  case AF_CONTROL_SS_FREQ | AF_CONTROL_GET:
-    *(float*)arg = s->freq;
-    return AF_OK;
-  case AF_CONTROL_SS_DECAY | AF_CONTROL_SET:
-    s->decay = *(float*)arg;
-    return AF_OK;
-  case AF_CONTROL_SS_DECAY | AF_CONTROL_GET:
-    *(float*)arg = s->decay;
-    return AF_OK;
   }
   return AF_UNKNOWN;
 }
diff --git a/audio/filter/af_sub.c b/audio/filter/af_sub.c
index 4af28d9141..a985ac2a05 100644
--- a/audio/filter/af_sub.c
+++ b/audio/filter/af_sub.c
@@ -70,9 +70,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = max(s->ch+1,((struct mp_audio*)arg)->nch);
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
+    mp_audio_set_channels_old(af->data, max(s->ch+1,((struct mp_audio*)arg)->nch));
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     // Design low-pass filter
     s->k = 1.0;
diff --git a/audio/filter/af_surround.c b/audio/filter/af_surround.c
index 57288d6ba2..c04a039d65 100644
--- a/audio/filter/af_surround.c
+++ b/audio/filter/af_surround.c
@@ -92,10 +92,9 @@ static int control(struct af_instance* af, int cmd, void* arg)
   switch(cmd){
   case AF_CONTROL_REINIT:{
     float fc;
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch*2;
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch*2);
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     if (af->data->nch != 4){
       mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Only stereo input is supported.\n");
@@ -125,8 +124,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
     if((af->data->format != ((struct mp_audio*)arg)->format) ||
        (af->data->bps    != ((struct mp_audio*)arg)->bps)){
-      ((struct mp_audio*)arg)->format = af->data->format;
-      ((struct mp_audio*)arg)->bps = af->data->bps;
+      mp_audio_set_format((struct mp_audio*)arg, af->data->format);
       return AF_FALSE;
     }
     return AF_OK;
@@ -244,7 +242,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data){
   // Set output data
   data->audio = af->data->audio;
   data->len   *= 2;
-  data->nch   = af->data->nch;
+  mp_audio_set_channels_old(data, af->data->nch);
 
   return data;
 }
diff --git a/audio/filter/af_sweep.c b/audio/filter/af_sweep.c
index 6d1106fefc..6cc099f2d8 100644
--- a/audio/filter/af_sweep.c
+++ b/audio/filter/af_sweep.c
@@ -41,12 +41,10 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
   switch(cmd){
   case AF_CONTROL_REINIT:
-    af->data->nch    = data->nch;
-    af->data->format = AF_FORMAT_S16_NE;
-    af->data->bps    = 2;
-    af->data->rate   = data->rate;
+    mp_audio_copy_config(af->data, data);
+    mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 
-    return AF_OK;
+    return af_test_output(af, data);
   case AF_CONTROL_COMMAND_LINE:
     sscanf((char*)arg,"%lf", &s->delta);
     return AF_OK;
diff --git a/audio/filter/af_tools.c b/audio/filter/af_tools.c
index 0d5dc6c573..77fdad55f2 100644
--- a/audio/filter/af_tools.c
+++ b/audio/filter/af_tools.c
@@ -90,8 +90,8 @@ int af_test_output(struct af_instance* af, struct mp_audio* out)
   if((af->data->format != out->format) ||
      (af->data->bps    != out->bps)    ||
      (af->data->rate   != out->rate)   ||
-     (af->data->nch    != out->nch)){
-    memcpy(out,af->data,sizeof(struct mp_audio));
+     !mp_chmap_equals(&af->data->channels, &out->channels)){
+    *out = *af->data;
     return AF_FALSE;
   }
   return AF_OK;
diff --git a/audio/filter/af_volume.c b/audio/filter/af_volume.c
index ecf181c8b8..82c31eaa12 100644
--- a/audio/filter/af_volume.c
+++ b/audio/filter/af_volume.c
@@ -66,12 +66,10 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
 
     if(s->fast && (((struct mp_audio*)arg)->format != (AF_FORMAT_FLOAT_NE))){
-      af->data->format = AF_FORMAT_S16_NE;
-      af->data->bps    = 2;
+      mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
     }
     else{
       // Cutoff set to 10Hz for forgetting factor
@@ -79,42 +77,21 @@ static int control(struct af_instance* af, int cmd, void* arg)
       float t = 2.0-cos(x);
       s->time = 1.0 - (t - sqrt(t*t - 1));
       mp_msg(MSGT_AFILTER, MSGL_DBG2, "[volume] Forgetting factor = %0.5f\n",s->time);
-      af->data->format = AF_FORMAT_FLOAT_NE;
-      af->data->bps    = 4;
+      mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
     }
     return af_test_output(af,(struct mp_audio*)arg);
   case AF_CONTROL_COMMAND_LINE:{
     float v=0.0;
     float vol[AF_NCH];
     int   i;
-    sscanf((char*)arg,"%f:%i", &v, &s->soft);
+    sscanf((char*)arg,"%f:%i:%i", &v, &s->soft, &s->fast);
     for(i=0;i<AF_NCH;i++) vol[i]=v;
     return control(af,AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET, vol);
   }
-  case AF_CONTROL_POST_CREATE:
-    s->fast = ((((struct af_cfg*)arg)->force & AF_INIT_FORMAT_MASK) ==
-      AF_INIT_FLOAT) ? 0 : 1;
-    return AF_OK;
-  case AF_CONTROL_VOLUME_ON_OFF | AF_CONTROL_SET:
-    memcpy(s->enable,(int*)arg,AF_NCH*sizeof(int));
-    return AF_OK;
-  case AF_CONTROL_VOLUME_ON_OFF | AF_CONTROL_GET:
-    memcpy((int*)arg,s->enable,AF_NCH*sizeof(int));
-    return AF_OK;
-  case AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_SET:
-    s->soft = *(int*)arg;
-    return AF_OK;
-  case AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_GET:
-    *(int*)arg = s->soft;
-    return AF_OK;
   case AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET:
     return af_from_dB(AF_NCH,(float*)arg,s->level,20.0,-200.0,60.0);
   case AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_GET:
     return af_to_dB(AF_NCH,s->level,(float*)arg,20.0);
-  case AF_CONTROL_VOLUME_PROBE | AF_CONTROL_GET:
-    return af_to_dB(AF_NCH,s->pow,(float*)arg,10.0);
-  case AF_CONTROL_VOLUME_PROBE_MAX | AF_CONTROL_GET:
-    return af_to_dB(AF_NCH,s->max,(float*)arg,10.0);
   case AF_CONTROL_PRE_DESTROY:{
     float m = 0.0;
     int i;
@@ -122,7 +99,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
       for(i=0;i<AF_NCH;i++)
 	m=max(m,s->max[i]);
 	af_to_dB(1, &m, &m, 10.0);
-	mp_msg(MSGT_AFILTER, MSGL_INFO, "[volume] The maximum volume was %0.2fdB \n", m);
+	mp_msg(MSGT_AFILTER, MSGL_V, "[volume] The maximum volume was %0.2fdB \n", m);
     }
     return AF_OK;
   }
diff --git a/audio/filter/control.h b/audio/filter/control.h
index 323b9a3924..aa1900d106 100644
--- a/audio/filter/control.h
+++ b/audio/filter/control.h
@@ -22,48 +22,6 @@
 #include <sys/types.h>
 
 /*********************************************
-// Control info struct.
-//
-// This struct is the argument in a info call to a filter.
-*/
-
-// Argument types
-#define AF_CONTROL_TYPE_BOOL	(0x0<<0)
-#define AF_CONTROL_TYPE_CHAR	(0x1<<0)
-#define AF_CONTROL_TYPE_INT	(0x2<<0)
-#define AF_CONTROL_TYPE_FLOAT	(0x3<<0)
-#define AF_CONTROL_TYPE_STRUCT	(0x4<<0)
-#define AF_CONTROL_TYPE_SPECIAL	(0x5<<0) // a pointer to a function for example
-#define AF_CONTROL_TYPE_MASK	(0x7<<0)
-// Argument geometry
-#define AF_CONTROL_GEOM_SCALAR	(0x0<<3)
-#define AF_CONTROL_GEOM_ARRAY	(0x1<<3)
-#define AF_CONTROL_GEOM_MATRIX	(0x2<<3)
-#define AF_CONTROL_GEOM_MASK	(0x3<<3)
-// Argument properties
-#define AF_CONTROL_PROP_READ	(0x0<<5) // The argument can be read
-#define AF_CONTROL_PROP_WRITE	(0x1<<5) // The argument can be written
-#define AF_CONTROL_PROP_SAVE	(0x2<<5) // Can be saved
-#define AF_CONTROL_PROP_RUNTIME	(0x4<<5) // Acessable during execution
-#define AF_CONTROL_PROP_CHANNEL (0x8<<5) // Argument is set per channel
-#define AF_CONTROL_PROP_MASK	(0xF<<5)
-
-typedef struct af_control_info_s{
-  int	 def;	// Control enumrification
-  char*	 name; 	// Name of argument
-  char*	 info;	// Description of what it does
-  int 	 flags;	// Flags as defined above
-  float	 max;	// Max and min value
-  float	 min;	// (only aplicable on float and int)
-  int	 xdim;	// 1st dimension
-  int	 ydim;	// 2nd dimension (=0 for everything except matrix)
-  size_t sz;	// Size of argument in bytes
-  int	 ch;	// Channel number (for future use)
-  void*  arg;	// Data (for future use)
-}af_control_info_t;
-
-
-/*********************************************
 // Extended control used with arguments that operates on only one
 // channel at the time
 */
@@ -98,11 +56,6 @@ typedef struct af_control_ext_s{
 
 // OPTIONAL CALLS
 
-/* Called just after creation with the af_cfg for the stream in which
-   the filter resides as input parameter this call can be used by the
-   filter to initialize itself */
-#define AF_CONTROL_POST_CREATE 		0x00000100 | AF_CONTROL_OPTIONAL
-
 // Called just before destruction of a filter
 #define AF_CONTROL_PRE_DESTROY 		0x00000200 | AF_CONTROL_OPTIONAL
 
@@ -119,20 +72,12 @@ typedef struct af_control_ext_s{
 #define AF_CONTROL_SET			0x00000000
 // Get argument
 #define AF_CONTROL_GET			0x00000001
-// Get info about the control, i.e fill in everything except argument
-#define AF_CONTROL_INFO			0x00000002
 
 // Resample
 
 // Set output rate in resample
 #define AF_CONTROL_RESAMPLE_RATE	0x00000100 | AF_CONTROL_FILTER_SPECIFIC
 
-// Enable sloppy resampling
-#define AF_CONTROL_RESAMPLE_SLOPPY	0x00000200 | AF_CONTROL_FILTER_SPECIFIC
-
-// Set resampling accuracy
-#define AF_CONTROL_RESAMPLE_ACCURACY	0x00000300 | AF_CONTROL_FILTER_SPECIFIC
-
 // Format
 
 #define AF_CONTROL_FORMAT_FMT		0x00000400 | AF_CONTROL_FILTER_SPECIFIC
@@ -142,69 +87,11 @@ typedef struct af_control_ext_s{
 // Set number of output channels in channels
 #define AF_CONTROL_CHANNELS		0x00000600 | AF_CONTROL_FILTER_SPECIFIC
 
-// Set number of channel routes
-#define AF_CONTROL_CHANNELS_ROUTES	0x00000700 | AF_CONTROL_FILTER_SPECIFIC
-
-// Set channel routing pair, arg is int[2] and ch is used
-#define AF_CONTROL_CHANNELS_ROUTING	0x00000800 | AF_CONTROL_FILTER_SPECIFIC
-
-// Set nuber of channel routing pairs, arg is int*
-#define AF_CONTROL_CHANNELS_NR		0x00000900 | AF_CONTROL_FILTER_SPECIFIC
-
-// Set make af_channels into a router
-#define AF_CONTROL_CHANNELS_ROUTER	0x00000A00 | AF_CONTROL_FILTER_SPECIFIC
-
 // Volume
 
-// Turn volume control on and off, arg is int*
-#define AF_CONTROL_VOLUME_ON_OFF	0x00000B00 | AF_CONTROL_FILTER_SPECIFIC
-
-// Turn soft clipping of the volume on and off, arg is binary
-#define AF_CONTROL_VOLUME_SOFTCLIP	0x00000C00 | AF_CONTROL_FILTER_SPECIFIC
-
 // Set volume level, arg is a float* with the volume for all the channels
 #define AF_CONTROL_VOLUME_LEVEL		0x00000D00 | AF_CONTROL_FILTER_SPECIFIC
 
-// Probed power level for all channels, arg is a float*
-#define AF_CONTROL_VOLUME_PROBE		0x00000E00 | AF_CONTROL_FILTER_SPECIFIC
-
-// Maximum probed power level for all channels, arg is a float*
-#define AF_CONTROL_VOLUME_PROBE_MAX	0x00000F00 | AF_CONTROL_FILTER_SPECIFIC
-
-// Compressor/expander
-
-// Turn compressor/expander on and off
-#define AF_CONTROL_COMP_ON_OFF	 	0x00001000 | AF_CONTROL_FILTER_SPECIFIC
-
-// Compression/expansion threshold [dB]
-#define AF_CONTROL_COMP_THRESH	 	0x00001100 | AF_CONTROL_FILTER_SPECIFIC
-
-// Compression/expansion attack time [ms]
-#define AF_CONTROL_COMP_ATTACK	 	0x00001200 | AF_CONTROL_FILTER_SPECIFIC
-
-// Compression/expansion release time [ms]
-#define AF_CONTROL_COMP_RELEASE 	0x00001300 | AF_CONTROL_FILTER_SPECIFIC
-
-// Compression/expansion gain level [dB]
-#define AF_CONTROL_COMP_RATIO	 	0x00001400 | AF_CONTROL_FILTER_SPECIFIC
-
-// Noise gate
-
-// Turn noise gate on an off
-#define AF_CONTROL_GATE_ON_OFF	 	0x00001500 | AF_CONTROL_FILTER_SPECIFIC
-
-// Noise gate threshold [dB]
-#define AF_CONTROL_GATE_THRESH	 	0x00001600 | AF_CONTROL_FILTER_SPECIFIC
-
-// Noise gate attack time [ms]
-#define AF_CONTROL_GATE_ATTACK	 	0x00001700 | AF_CONTROL_FILTER_SPECIFIC
-
-// Noise gate release time [ms]
-#define AF_CONTROL_GATE_RELEASE 	0x00001800 | AF_CONTROL_FILTER_SPECIFIC
-
-// Noise gate release range level [dB]
-#define AF_CONTROL_GATE_RANGE	 	0x00001900 | AF_CONTROL_FILTER_SPECIFIC
-
 // Pan
 
 // Pan levels, arg is a control_ext with a float*
@@ -216,9 +103,6 @@ typedef struct af_control_ext_s{
 // Balance, arg is float*; range -1 (left) to 1 (right), 0 center
 #define AF_CONTROL_PAN_BALANCE	 	0x00001C00 | AF_CONTROL_FILTER_SPECIFIC
 
-// Set equalizer gain, arg is a control_ext with a float*
-#define AF_CONTROL_EQUALIZER_GAIN 	0x00001D00 | AF_CONTROL_FILTER_SPECIFIC
-
 
 // Delay length in ms, arg is a control_ext with a float*
 #define AF_CONTROL_DELAY_LEN		0x00001E00 | AF_CONTROL_FILTER_SPECIFIC
@@ -236,21 +120,10 @@ typedef struct af_control_ext_s{
 // Export
 #define AF_CONTROL_EXPORT_SZ            0x00003000 | AF_CONTROL_FILTER_SPECIFIC
 
-
-// ExtraStereo Multiplier
-#define AF_CONTROL_ES_MUL		0x00003100 | AF_CONTROL_FILTER_SPECIFIC
-
-
-// Center
-
 // Channel number which to inster the filtered data, arg in int*
 #define AF_CONTROL_CENTER_CH		0x00003200 | AF_CONTROL_FILTER_SPECIFIC
 
 
-// SineSuppress
-#define AF_CONTROL_SS_FREQ		0x00003300 | AF_CONTROL_FILTER_SPECIFIC
-#define AF_CONTROL_SS_DECAY		0x00003400 | AF_CONTROL_FILTER_SPECIFIC
-
 #define AF_CONTROL_PLAYBACK_SPEED	0x00003500 | AF_CONTROL_FILTER_SPECIFIC
 #define AF_CONTROL_SCALETEMPO_AMOUNT	0x00003600 | AF_CONTROL_FILTER_SPECIFIC
 
diff --git a/audio/fmt-conversion.c b/audio/fmt-conversion.c
new file mode 100644
index 0000000000..4c1055f118
--- /dev/null
+++ b/audio/fmt-conversion.c
@@ -0,0 +1,65 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */ 
+
+#include "core/mp_msg.h"
+#include <libavutil/avutil.h>
+#include <libavutil/samplefmt.h>
+#include "format.h"
+#include "fmt-conversion.h"
+
+static const struct {
+    enum AVSampleFormat sample_fmt;
+    int fmt;
+} audio_conversion_map[] = {
+    {AV_SAMPLE_FMT_U8,    AF_FORMAT_U8},
+    {AV_SAMPLE_FMT_S16,   AF_FORMAT_S16_NE},
+    {AV_SAMPLE_FMT_S32,   AF_FORMAT_S32_NE},
+    {AV_SAMPLE_FMT_FLT,   AF_FORMAT_FLOAT_NE},
+    {AV_SAMPLE_FMT_DBL,   AF_FORMAT_DOUBLE_NE},
+
+    {AV_SAMPLE_FMT_NONE,  0},
+};
+
+enum AVSampleFormat af_to_avformat(int fmt)
+{
+    int i;
+    enum AVSampleFormat sample_fmt;
+    for (i = 0; audio_conversion_map[i].fmt; i++)
+        if (audio_conversion_map[i].fmt == fmt)
+            break;
+    sample_fmt = audio_conversion_map[i].sample_fmt;
+    if (sample_fmt == AF_FORMAT_UNKNOWN)
+        mp_msg(MSGT_GLOBAL, MSGL_V, "Unsupported sample format: %s\n",
+                af_fmt2str_short(fmt));
+    return sample_fmt;
+}
+
+int af_from_avformat(enum AVSampleFormat sample_fmt)
+{
+    int i;
+    for (i = 0; audio_conversion_map[i].fmt; i++)
+        if (audio_conversion_map[i].sample_fmt == sample_fmt)
+            break;
+    int fmt = audio_conversion_map[i].fmt;
+    if (!fmt) {
+        const char *fmtname = av_get_sample_fmt_name(sample_fmt);
+        mp_msg(MSGT_GLOBAL, MSGL_ERR, "Unsupported AVSampleFormat %s (%d)\n",
+               fmtname ? fmtname : "INVALID", sample_fmt);
+    }
+    return fmt;
+}
diff --git a/audio/fmt-conversion.h b/audio/fmt-conversion.h
new file mode 100644
index 0000000000..7f2739f86f
--- /dev/null
+++ b/audio/fmt-conversion.h
@@ -0,0 +1,25 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPLAYER_SAMPLE_FMT_CONVERSION_H
+#define MPLAYER_SAMPLE_FMT_CONVERSION_H
+
+enum AVSampleFormat af_to_avformat(int fmt);
+int af_from_avformat(enum AVSampleFormat sample_fmt);
+
+#endif /* MPLAYER_SAMPLE_FMT_CONVERSION_H */
diff --git a/audio/format.c b/audio/format.c
index 9625857ada..5b1262956c 100644
--- a/audio/format.c
+++ b/audio/format.c
@@ -29,24 +29,17 @@
 int af_fmt2bits(int format)
 {
     if (AF_FORMAT_IS_AC3(format)) return 16;
-    return (format & AF_FORMAT_BITS_MASK)+8;
-//    return (((format & AF_FORMAT_BITS_MASK)>>3)+1) * 8;
-#if 0
+    if (format == AF_FORMAT_UNKNOWN)
+        return 0;
     switch(format & AF_FORMAT_BITS_MASK)
     {
 	case AF_FORMAT_8BIT: return 8;
 	case AF_FORMAT_16BIT: return 16;
 	case AF_FORMAT_24BIT: return 24;
 	case AF_FORMAT_32BIT: return 32;
-	case AF_FORMAT_48BIT: return 48;
+	case AF_FORMAT_64BIT: return 64;
     }
-#endif
-    return -1;
-}
-
-int af_bits2fmt(int bits)
-{
-    return (bits/8 - 1) << 3;
+    return 0;
 }
 
 /* Convert format to str input str is a buffer for the
@@ -94,6 +87,9 @@ const struct af_fmt_entry af_fmtstr_table[] = {
     { "floatle", AF_FORMAT_FLOAT_LE },
     { "floatbe", AF_FORMAT_FLOAT_BE },
     { "floatne", AF_FORMAT_FLOAT_NE },
+    { "doublele", AF_FORMAT_DOUBLE_LE },
+    { "doublebe", AF_FORMAT_DOUBLE_BE },
+    { "doublene", AF_FORMAT_DOUBLE_NE },
 
     {0}
 };
diff --git a/audio/format.h b/audio/format.h
index a8249954f0..30a4aa1cea 100644
--- a/audio/format.h
+++ b/audio/format.h
@@ -53,8 +53,7 @@
 #define AF_FORMAT_16BIT		(1<<3)
 #define AF_FORMAT_24BIT		(2<<3)
 #define AF_FORMAT_32BIT		(3<<3)
-#define AF_FORMAT_40BIT		(4<<3)
-#define AF_FORMAT_48BIT		(5<<3)
+#define AF_FORMAT_64BIT		(4<<3)
 #define AF_FORMAT_BITS_MASK	(7<<3)
 
 // Special flags refering to non pcm data (note: 1<<6, 2<<6, 5<<6 unused)
@@ -85,6 +84,9 @@
 #define AF_FORMAT_FLOAT_LE	(AF_FORMAT_F|AF_FORMAT_32BIT|AF_FORMAT_LE)
 #define AF_FORMAT_FLOAT_BE	(AF_FORMAT_F|AF_FORMAT_32BIT|AF_FORMAT_BE)
 
+#define AF_FORMAT_DOUBLE_LE     (AF_FORMAT_F|AF_FORMAT_64BIT|AF_FORMAT_LE)
+#define AF_FORMAT_DOUBLE_BE     (AF_FORMAT_F|AF_FORMAT_64BIT|AF_FORMAT_BE)
+
 #define AF_FORMAT_AC3_LE	(AF_FORMAT_AC3|AF_FORMAT_16BIT|AF_FORMAT_LE)
 #define AF_FORMAT_AC3_BE	(AF_FORMAT_AC3|AF_FORMAT_16BIT|AF_FORMAT_BE)
 
@@ -99,6 +101,7 @@
 #define AF_FORMAT_U32_NE AF_FORMAT_U32_BE
 #define AF_FORMAT_S32_NE AF_FORMAT_S32_BE
 #define AF_FORMAT_FLOAT_NE AF_FORMAT_FLOAT_BE
+#define AF_FORMAT_DOUBLE_NE AF_FORMAT_DOUBLE_BE
 #define AF_FORMAT_AC3_NE AF_FORMAT_AC3_BE
 #define AF_FORMAT_IEC61937_NE AF_FORMAT_IEC61937_BE
 #else
@@ -109,6 +112,7 @@
 #define AF_FORMAT_U32_NE AF_FORMAT_U32_LE
 #define AF_FORMAT_S32_NE AF_FORMAT_S32_LE
 #define AF_FORMAT_FLOAT_NE AF_FORMAT_FLOAT_LE
+#define AF_FORMAT_DOUBLE_NE AF_FORMAT_DOUBLE_LE
 #define AF_FORMAT_AC3_NE AF_FORMAT_AC3_LE
 #define AF_FORMAT_IEC61937_NE AF_FORMAT_IEC61937_LE
 #endif
@@ -127,7 +131,6 @@ extern const struct af_fmt_entry af_fmtstr_table[];
 
 int af_str2fmt_short(bstr str);
 int af_fmt2bits(int format);
-int af_bits2fmt(int bits);
 char* af_fmt2str(int format, char* str, int size);
 const char* af_fmt2str_short(int format);
 
diff --git a/audio/mixer.c b/audio/mixer.c
index b67e548a3c..0f1a7871d6 100644
--- a/audio/mixer.c
+++ b/audio/mixer.c
@@ -217,7 +217,7 @@ void mixer_setbalance(mixer_t *mixer, float val)
                            AF_CONTROL_PAN_BALANCE | AF_CONTROL_SET, &val))
         return;
 
-    if (val == 0 || mixer->ao->channels < 2)
+    if (val == 0 || mixer->ao->channels.num < 2)
         return;
 
     if (!(af_pan_balance = af_add(mixer->afilter, "pan"))) {
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);
diff --git a/audio/reorder_ch.c b/audio/reorder_ch.c
index 72849c3d13..3a2b747668 100644
--- a/audio/reorder_ch.c
+++ b/audio/reorder_ch.c
@@ -20,1384 +20,11 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include <stdio.h>
-#include <stdlib.h>
 #include <inttypes.h>
 #include <string.h>
 
 #include "audio/reorder_ch.h"
 
-#ifdef TEST
-#define mp_msg(mod,lev, fmt, args... )  printf( fmt, ## args )
-#else
-#include "core/mp_msg.h"
-#endif
-
-
-#define REORDER_COPY_5(DEST,SRC,SAMPLES,S0,S1,S2,S3,S4) \
-for (i = 0; i < SAMPLES; i += 5) {\
-    DEST[i]   = SRC[i+S0];\
-    DEST[i+1] = SRC[i+S1];\
-    DEST[i+2] = SRC[i+S2];\
-    DEST[i+3] = SRC[i+S3];\
-    DEST[i+4] = SRC[i+S4];\
-}
-
-static int reorder_copy_5ch(void *dest, const void *src,
-                            unsigned int samples, unsigned int samplesize,
-                            int s0, int s1, int s2, int s3, int s4)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *dest_8 = dest;
-        const int8_t *src_8 = src;
-        REORDER_COPY_5(dest_8,src_8,samples,s0,s1,s2,s3,s4);
-        break;
-    }
-    case 2:
-    {
-        int16_t *dest_16 = dest;
-        const int16_t *src_16 = src;
-        REORDER_COPY_5(dest_16,src_16,samples,s0,s1,s2,s3,s4);
-        break;
-    }
-    case 3:
-    {
-        int8_t *dest_8 = dest;
-        const int8_t *src_8 = src;
-        for (i = 0; i < samples * 3; i += 15) {
-            dest_8[i]    = src_8[i+s0*3];
-            dest_8[i+1]  = src_8[i+s0*3+1];
-            dest_8[i+2]  = src_8[i+s0*3+2];
-            dest_8[i+3]  = src_8[i+s1*3];
-            dest_8[i+4]  = src_8[i+s1*3+1];
-            dest_8[i+5]  = src_8[i+s1*3+2];
-            dest_8[i+6]  = src_8[i+s2*3];
-            dest_8[i+7]  = src_8[i+s2*3+1];
-            dest_8[i+8]  = src_8[i+s2*3+2];
-            dest_8[i+9]  = src_8[i+s3*3];
-            dest_8[i+10] = src_8[i+s3*3+1];
-            dest_8[i+11] = src_8[i+s3*3+2];
-            dest_8[i+12] = src_8[i+s4*3];
-            dest_8[i+13] = src_8[i+s4*3+1];
-            dest_8[i+14] = src_8[i+s4*3+2];
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *dest_32 = dest;
-        const int32_t *src_32 = src;
-        REORDER_COPY_5(dest_32,src_32,samples,s0,s1,s2,s3,s4);
-        break;
-    }
-    case 8:
-    {
-        int64_t *dest_64 = dest;
-        const int64_t *src_64 = src;
-        REORDER_COPY_5(dest_64,src_64,samples,s0,s1,s2,s3,s4);
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-#define REORDER_COPY_6(DEST,SRC,SAMPLES,S0,S1,S2,S3,S4,S5) \
-for (i = 0; i < SAMPLES; i += 6) {\
-    DEST[i]   = SRC[i+S0];\
-    DEST[i+1] = SRC[i+S1];\
-    DEST[i+2] = SRC[i+S2];\
-    DEST[i+3] = SRC[i+S3];\
-    DEST[i+4] = SRC[i+S4];\
-    DEST[i+5] = SRC[i+S5];\
-}
-
-static int reorder_copy_6ch(void *dest, const void *src,
-                            unsigned int samples, uint8_t samplesize,
-                            int s0, int s1, int s2, int s3, int s4, int s5)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *dest_8 = dest;
-        const int8_t *src_8 = src;
-        REORDER_COPY_6(dest_8,src_8,samples,s0,s1,s2,s3,s4,s5);
-        break;
-    }
-    case 2:
-    {
-        int16_t *dest_16 = dest;
-        const int16_t *src_16 = src;
-        REORDER_COPY_6(dest_16,src_16,samples,s0,s1,s2,s3,s4,s5);
-        break;
-    }
-    case 3:
-    {
-        int8_t *dest_8 = dest;
-        const int8_t *src_8 = src;
-        for (i = 0; i < samples * 3; i += 18) {
-            dest_8[i]    = src_8[i+s0*3];
-            dest_8[i+1]  = src_8[i+s0*3+1];
-            dest_8[i+2]  = src_8[i+s0*3+2];
-            dest_8[i+3]  = src_8[i+s1*3];
-            dest_8[i+4]  = src_8[i+s1*3+1];
-            dest_8[i+5]  = src_8[i+s1*3+2];
-            dest_8[i+6]  = src_8[i+s2*3];
-            dest_8[i+7]  = src_8[i+s2*3+1];
-            dest_8[i+8]  = src_8[i+s2*3+2];
-            dest_8[i+9]  = src_8[i+s3*3];
-            dest_8[i+10] = src_8[i+s3*3+1];
-            dest_8[i+11] = src_8[i+s3*3+2];
-            dest_8[i+12] = src_8[i+s4*3];
-            dest_8[i+13] = src_8[i+s4*3+1];
-            dest_8[i+14] = src_8[i+s4*3+2];
-            dest_8[i+15] = src_8[i+s5*3];
-            dest_8[i+16] = src_8[i+s5*3+1];
-            dest_8[i+17] = src_8[i+s5*3+2];
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *dest_32 = dest;
-        const int32_t *src_32 = src;
-        REORDER_COPY_6(dest_32,src_32,samples,s0,s1,s2,s3,s4,s5);
-        break;
-    }
-    case 8:
-    {
-        int64_t *dest_64 = dest;
-        const int64_t *src_64 = src;
-        REORDER_COPY_6(dest_64,src_64,samples,s0,s1,s2,s3,s4,s5);
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-#define REORDER_COPY_8(DEST,SRC,SAMPLES,S0,S1,S2,S3,S4,S5,S6,S7) \
-for (i = 0; i < SAMPLES; i += 8) {\
-    DEST[i]   = SRC[i+S0];\
-    DEST[i+1] = SRC[i+S1];\
-    DEST[i+2] = SRC[i+S2];\
-    DEST[i+3] = SRC[i+S3];\
-    DEST[i+4] = SRC[i+S4];\
-    DEST[i+5] = SRC[i+S5];\
-    DEST[i+6] = SRC[i+S6];\
-    DEST[i+7] = SRC[i+S7];\
-}
-
-static int reorder_copy_8ch(void *dest, const void *src,
-                            unsigned int samples, uint8_t samplesize,
-                            int s0, int s1, int s2, int s3,
-                            int s4, int s5, int s6, int s7)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *dest_8 = dest;
-        const int8_t *src_8 = src;
-        REORDER_COPY_8(dest_8,src_8,samples,s0,s1,s2,s3,s4,s5,s6,s7);
-        break;
-    }
-    case 2:
-    {
-        int16_t *dest_16 = dest;
-        const int16_t *src_16 = src;
-        REORDER_COPY_8(dest_16,src_16,samples,s0,s1,s2,s3,s4,s5,s6,s7);
-        break;
-    }
-    case 3:
-    {
-        int8_t *dest_8 = dest;
-        const int8_t *src_8 = src;
-        for (i = 0; i < samples * 3; i += 24) {
-            dest_8[i]    = src_8[i+s0*3];
-            dest_8[i+1]  = src_8[i+s0*3+1];
-            dest_8[i+2]  = src_8[i+s0*3+2];
-            dest_8[i+3]  = src_8[i+s1*3];
-            dest_8[i+4]  = src_8[i+s1*3+1];
-            dest_8[i+5]  = src_8[i+s1*3+2];
-            dest_8[i+6]  = src_8[i+s2*3];
-            dest_8[i+7]  = src_8[i+s2*3+1];
-            dest_8[i+8]  = src_8[i+s2*3+2];
-            dest_8[i+9]  = src_8[i+s3*3];
-            dest_8[i+10] = src_8[i+s3*3+1];
-            dest_8[i+11] = src_8[i+s3*3+2];
-            dest_8[i+12] = src_8[i+s4*3];
-            dest_8[i+13] = src_8[i+s4*3+1];
-            dest_8[i+14] = src_8[i+s4*3+2];
-            dest_8[i+15] = src_8[i+s5*3];
-            dest_8[i+16] = src_8[i+s5*3+1];
-            dest_8[i+17] = src_8[i+s5*3+2];
-            dest_8[i+18] = src_8[i+s6*3];
-            dest_8[i+19] = src_8[i+s6*3+1];
-            dest_8[i+20] = src_8[i+s6*3+2];
-            dest_8[i+21] = src_8[i+s7*3];
-            dest_8[i+22] = src_8[i+s7*3+1];
-            dest_8[i+23] = src_8[i+s7*3+2];
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *dest_32 = dest;
-        const int32_t *src_32 = src;
-        REORDER_COPY_8(dest_32,src_32,samples,s0,s1,s2,s3,s4,s5,s6,s7);
-        break;
-    }
-    case 8:
-    {
-        int64_t *dest_64 = dest;
-        const int64_t *src_64 = src;
-        REORDER_COPY_8(dest_64,src_64,samples,s0,s1,s2,s3,s4,s5,s6,s7);
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-void reorder_channel_copy(void *src,
-                          int src_layout,
-                          void *dest,
-                          int dest_layout,
-                          int samples,
-                          int samplesize)
-{
-    if (dest_layout==src_layout) {
-        memcpy(dest, src, samples*samplesize);
-        return;
-    }
-    if (!AF_IS_SAME_CH_NUM(dest_layout,src_layout)) {
-        mp_msg(MSGT_GLOBAL, MSGL_WARN, "[reorder_ch] different channel count "
-               "between src and dest: %x, %x\n",
-               AF_GET_CH_NUM_WITH_LFE(src_layout),
-               AF_GET_CH_NUM_WITH_LFE(dest_layout));
-        return;
-    }
-    switch ((src_layout<<16)|dest_layout) {
-    // AF_CHANNEL_LAYOUT_5_0_A   L R C Ls Rs
-    // AF_CHANNEL_LAYOUT_5_0_B   L R Ls Rs C
-    // AF_CHANNEL_LAYOUT_5_0_C   L C R Ls Rs
-    // AF_CHANNEL_LAYOUT_5_0_D   C L R Ls Rs
-    case AF_CHANNEL_LAYOUT_5_0_A << 16 | AF_CHANNEL_LAYOUT_5_0_B:
-        reorder_copy_5ch(dest, src, samples, samplesize, 0, 1, 3, 4, 2);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_A << 16 | AF_CHANNEL_LAYOUT_5_0_C:
-        reorder_copy_5ch(dest, src, samples, samplesize, 0, 2, 1, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_A << 16 | AF_CHANNEL_LAYOUT_5_0_D:
-        reorder_copy_5ch(dest, src, samples, samplesize, 2, 0, 1, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_B << 16 | AF_CHANNEL_LAYOUT_5_0_A:
-        reorder_copy_5ch(dest, src, samples, samplesize, 0, 1, 4, 2, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_B << 16 | AF_CHANNEL_LAYOUT_5_0_C:
-        reorder_copy_5ch(dest, src, samples, samplesize, 0, 4, 1, 2, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_B << 16 | AF_CHANNEL_LAYOUT_5_0_D:
-        reorder_copy_5ch(dest, src, samples, samplesize, 4, 0, 1, 2, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_C << 16 | AF_CHANNEL_LAYOUT_5_0_A:
-        reorder_copy_5ch(dest, src, samples, samplesize, 0, 2, 1, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_C << 16 | AF_CHANNEL_LAYOUT_5_0_B:
-        reorder_copy_5ch(dest, src, samples, samplesize, 0, 2, 3, 4, 1);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_C << 16 | AF_CHANNEL_LAYOUT_5_0_D:
-        reorder_copy_5ch(dest, src, samples, samplesize, 1, 0, 2, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_D << 16 | AF_CHANNEL_LAYOUT_5_0_A:
-        reorder_copy_5ch(dest, src, samples, samplesize, 1, 2, 0, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_D << 16 | AF_CHANNEL_LAYOUT_5_0_B:
-        reorder_copy_5ch(dest, src, samples, samplesize, 1, 2, 3, 4, 0);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_D << 16 | AF_CHANNEL_LAYOUT_5_0_C:
-        reorder_copy_5ch(dest, src, samples, samplesize, 1, 0, 2, 3, 4);
-        break;
-    // AF_CHANNEL_LAYOUT_5_1_A   L R C LFE Ls Rs
-    // AF_CHANNEL_LAYOUT_5_1_B   L R Ls Rs C LFE
-    // AF_CHANNEL_LAYOUT_5_1_C   L C R Ls Rs LFE
-    // AF_CHANNEL_LAYOUT_5_1_D   C L R Ls Rs LFE
-    // AF_CHANNEL_LAYOUT_5_1_E   LFE L C R Ls Rs
-    case AF_CHANNEL_LAYOUT_5_1_A << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        reorder_copy_6ch(dest, src, samples, samplesize, 0, 1, 4, 5, 2, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_A << 16 | AF_CHANNEL_LAYOUT_5_1_C:
-        reorder_copy_6ch(dest, src, samples, samplesize, 0, 2, 1, 4, 5, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_A << 16 | AF_CHANNEL_LAYOUT_5_1_D:
-        reorder_copy_6ch(dest, src, samples, samplesize, 2, 0, 1, 4, 5, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_B << 16 | AF_CHANNEL_LAYOUT_5_1_A:
-        reorder_copy_6ch(dest, src, samples, samplesize, 0, 1, 4, 5, 2, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_B << 16 | AF_CHANNEL_LAYOUT_5_1_C:
-        reorder_copy_6ch(dest, src, samples, samplesize, 0, 4, 1, 2, 3, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_B << 16 | AF_CHANNEL_LAYOUT_5_1_D:
-        reorder_copy_6ch(dest, src, samples, samplesize, 4, 0, 1, 2, 3, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_B << 16 | AF_CHANNEL_LAYOUT_5_1_E:
-        reorder_copy_6ch(dest, src, samples, samplesize, 5, 0, 4, 1, 2, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_C << 16 | AF_CHANNEL_LAYOUT_5_1_A:
-        reorder_copy_6ch(dest, src, samples, samplesize, 0, 2, 1, 5, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_C << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        reorder_copy_6ch(dest, src, samples, samplesize, 0, 2, 3, 4, 1, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_C << 16 | AF_CHANNEL_LAYOUT_5_1_D:
-        reorder_copy_6ch(dest, src, samples, samplesize, 1, 0, 2, 3, 4, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_D << 16 | AF_CHANNEL_LAYOUT_5_1_A:
-        reorder_copy_6ch(dest, src, samples, samplesize, 1, 2, 0, 5, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_D << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        reorder_copy_6ch(dest, src, samples, samplesize, 1, 2, 3, 4, 0, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_D << 16 | AF_CHANNEL_LAYOUT_5_1_C:
-        reorder_copy_6ch(dest, src, samples, samplesize, 1, 0, 2, 3, 4, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_E << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        reorder_copy_6ch(dest, src, samples, samplesize, 1, 3, 4, 5, 2, 0);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_F << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        reorder_copy_6ch(dest, src, samples, samplesize, 1, 2, 4, 5, 0, 3);
-        break;
-    // AF_CHANNEL_LAYOUT_7_1_A   L R C LFE Ls Rs Rls Rrs
-    // AF_CHANNEL_LAYOUT_7_1_B   L R Ls Rs C LFE Rls Rrs
-    // AF_CHANNEL_LAYOUT_7_1_D   C L R Ls Rs Rls Rrs LFE
-    case AF_CHANNEL_LAYOUT_7_1_A << 16 | AF_CHANNEL_LAYOUT_7_1_B:
-    case AF_CHANNEL_LAYOUT_7_1_B << 16 | AF_CHANNEL_LAYOUT_7_1_A:
-        reorder_copy_8ch(dest, src, samples, samplesize, 0, 1, 4, 5, 2, 3, 6, 7);
-        break;
-    case AF_CHANNEL_LAYOUT_7_1_D << 16 | AF_CHANNEL_LAYOUT_7_1_B:
-        reorder_copy_8ch(dest, src, samples, samplesize, 1, 2, 3, 4, 0, 7, 5, 6);
-        break;
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN, "[reorder_channel_copy] unsupport "
-               "from %x to %x, %d * %d\n", src_layout, dest_layout,
-               samples, samplesize);
-        memcpy(dest, src, samples*samplesize);
-    }
-}
-
-
-#define REORDER_SELF_SWAP_2(SRC,TMP,SAMPLES,CHNUM,S0,S1) \
-for (i = 0; i < SAMPLES; i += CHNUM) {\
-    TMP = SRC[i+S0];\
-    SRC[i+S0] = SRC[i+S1];\
-    SRC[i+S1] = TMP;\
-}
-
-static int reorder_self_2(void *src, unsigned int samples,
-                          unsigned int samplesize, unsigned int chnum,
-                          int s0, int s1)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_2(src_8,tmp,samples,6,s0,s1);
-        }
-        else if (chnum==8) {
-            REORDER_SELF_SWAP_2(src_8,tmp,samples,8,s0,s1);
-        }
-        else {
-            REORDER_SELF_SWAP_2(src_8,tmp,samples,5,s0,s1);
-        }
-        break;
-    }
-    case 2:
-    {
-        int16_t *src_16 = src;
-        int16_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_2(src_16,tmp,samples,6,s0,s1);
-        }
-        else if (chnum==3) {
-            REORDER_SELF_SWAP_2(src_16,tmp,samples,3,s0,s1);
-        }
-        else if (chnum==4) {
-            REORDER_SELF_SWAP_2(src_16,tmp,samples,3,s0,s1);
-        }
-        else {
-            REORDER_SELF_SWAP_2(src_16,tmp,samples,5,s0,s1);
-        }
-        break;
-    }
-    case 3:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp0, tmp1, tmp2;
-        for (i = 0; i < samples * 3; i += chnum * 3) {
-            tmp0 = src_8[i+s0*3];
-            tmp1 = src_8[i+s0*3+1];
-            tmp2 = src_8[i+s0*3+2];
-            src_8[i+s0*3]   = src_8[i+s1*3];
-            src_8[i+s0*3+1] = src_8[i+s1*3+1];
-            src_8[i+s0*3+2] = src_8[i+s1*3+2];
-            src_8[i+s1*3]   = tmp0;
-            src_8[i+s1*3+1] = tmp1;
-            src_8[i+s1*3+2] = tmp2;
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *src_32 = src;
-        int32_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_2(src_32,tmp,samples,6,s0,s1);
-        }
-        else if (chnum==3) {
-            REORDER_SELF_SWAP_2(src_32,tmp,samples,3,s0,s1);
-        }
-        else if (chnum==4) {
-            REORDER_SELF_SWAP_2(src_32,tmp,samples,4,s0,s1);
-        }
-        else {
-            REORDER_SELF_SWAP_2(src_32,tmp,samples,5,s0,s1);
-        }
-        break;
-    }
-    case 8:
-    {
-        int64_t *src_64 = src;
-        int64_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_2(src_64,tmp,samples,6,s0,s1);
-        }
-        else if (chnum==3) {
-            REORDER_SELF_SWAP_2(src_64,tmp,samples,3,s0,s1);
-        }
-        else if (chnum==4) {
-            REORDER_SELF_SWAP_2(src_64,tmp,samples,4,s0,s1);
-        }
-        else {
-            REORDER_SELF_SWAP_2(src_64,tmp,samples,5,s0,s1);
-        }
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-#define REORDER_SELF_SWAP_3(SRC,TMP,SAMPLES,CHNUM,S0,S1,S2) \
-for (i = 0; i < SAMPLES; i += CHNUM) {\
-    TMP = SRC[i+S0];\
-    SRC[i+S0] = SRC[i+S1];\
-    SRC[i+S1] = SRC[i+S2];\
-    SRC[i+S2] = TMP;\
-}
-
-static int reorder_self_3(void *src, unsigned int samples,
-                          unsigned int samplesize, unsigned int chnum,
-                          int s0, int s1, int s2)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_3(src_8,tmp,samples,6,s0,s1,s2);
-        }
-        else {
-            REORDER_SELF_SWAP_3(src_8,tmp,samples,5,s0,s1,s2);
-        }
-        break;
-    }
-    case 2:
-    {
-        int16_t *src_16 = src;
-        int16_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_3(src_16,tmp,samples,6,s0,s1,s2);
-        }
-        else {
-            REORDER_SELF_SWAP_3(src_16,tmp,samples,5,s0,s1,s2);
-        }
-        break;
-    }
-    case 3:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp0, tmp1, tmp2;
-        for (i = 0; i < samples * 3; i += chnum * 3) {
-            tmp0 = src_8[i+s0*3];
-            tmp1 = src_8[i+s0*3+1];
-            tmp2 = src_8[i+s0*3+2];
-            src_8[i+s0*3]   = src_8[i+s1*3];
-            src_8[i+s0*3+1] = src_8[i+s1*3+1];
-            src_8[i+s0*3+2] = src_8[i+s1*3+2];
-            src_8[i+s1*3]   = src_8[i+s2*3];
-            src_8[i+s1*3+1] = src_8[i+s2*3+1];
-            src_8[i+s1*3+2] = src_8[i+s2*3+2];
-            src_8[i+s2*3]   = tmp0;
-            src_8[i+s2*3+1] = tmp1;
-            src_8[i+s2*3+2] = tmp2;
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *src_32 = src;
-        int32_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_3(src_32,tmp,samples,6,s0,s1,s2);
-        }
-        else {
-            REORDER_SELF_SWAP_3(src_32,tmp,samples,5,s0,s1,s2);
-        }
-        break;
-    }
-    case 8:
-    {
-        int64_t *src_64 = src;
-        int64_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_3(src_64,tmp,samples,6,s0,s1,s2);
-        }
-        else {
-            REORDER_SELF_SWAP_3(src_64,tmp,samples,5,s0,s1,s2);
-        }
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-#define REORDER_SELF_SWAP_4_STEP_1(SRC,TMP,SAMPLES,CHNUM,S0,S1,S2,S3) \
-for (i = 0; i < SAMPLES; i += CHNUM) {\
-    TMP = SRC[i+S0];\
-    SRC[i+S0] = SRC[i+S1];\
-    SRC[i+S1] = SRC[i+S2];\
-    SRC[i+S2] = SRC[i+S3];\
-    SRC[i+S3] = TMP;\
-}
-
-static int reorder_self_4_step_1(void *src, unsigned int samples,
-                                 unsigned int samplesize, unsigned int chnum,
-                                 int s0, int s1, int s2, int s3)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_4_STEP_1(src_8,tmp,samples,6,s0,s1,s2,s3);
-        }
-        else if (chnum==8) {
-            REORDER_SELF_SWAP_4_STEP_1(src_8,tmp,samples,8,s0,s1,s2,s3);
-        }
-        else {
-            REORDER_SELF_SWAP_4_STEP_1(src_8,tmp,samples,5,s0,s1,s2,s3);
-        }
-        break;
-    }
-    case 2:
-    {
-        int16_t *src_16 = src;
-        int16_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_4_STEP_1(src_16,tmp,samples,6,s0,s1,s2,s3);
-        }
-        else if (chnum==8) {
-            REORDER_SELF_SWAP_4_STEP_1(src_16,tmp,samples,8,s0,s1,s2,s3);
-        }
-        else {
-            REORDER_SELF_SWAP_4_STEP_1(src_16,tmp,samples,5,s0,s1,s2,s3);
-        }
-        break;
-    }
-    case 3:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp0, tmp1, tmp2;
-        for (i = 0; i < samples * 3; i += chnum * 3) {
-            tmp0 = src_8[i+s0*3];
-            tmp1 = src_8[i+s0*3+1];
-            tmp2 = src_8[i+s0*3+2];
-            src_8[i+s0*3]   = src_8[i+s1*3];
-            src_8[i+s0*3+1] = src_8[i+s1*3+1];
-            src_8[i+s0*3+2] = src_8[i+s1*3+2];
-            src_8[i+s1*3]   = src_8[i+s2*3];
-            src_8[i+s1*3+1] = src_8[i+s2*3+1];
-            src_8[i+s1*3+2] = src_8[i+s2*3+2];
-            src_8[i+s2*3]   = src_8[i+s3*3];
-            src_8[i+s2*3+1] = src_8[i+s3*3+1];
-            src_8[i+s2*3+2] = src_8[i+s3*3+2];
-            src_8[i+s3*3]   = tmp0;
-            src_8[i+s3*3+1] = tmp1;
-            src_8[i+s3*3+2] = tmp2;
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *src_32 = src;
-        int32_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_4_STEP_1(src_32,tmp,samples,6,s0,s1,s2,s3);
-        }
-        else if (chnum==8) {
-            REORDER_SELF_SWAP_4_STEP_1(src_32,tmp,samples,8,s0,s1,s2,s3);
-        }
-        else {
-            REORDER_SELF_SWAP_4_STEP_1(src_32,tmp,samples,5,s0,s1,s2,s3);
-        }
-        break;
-    }
-    case 8:
-    {
-        int64_t *src_64 = src;
-        int64_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_4_STEP_1(src_64,tmp,samples,6,s0,s1,s2,s3);
-        }
-        else if (chnum==8) {
-            REORDER_SELF_SWAP_4_STEP_1(src_64,tmp,samples,8,s0,s1,s2,s3);
-        }
-        else {
-            REORDER_SELF_SWAP_4_STEP_1(src_64,tmp,samples,5,s0,s1,s2,s3);
-        }
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-#define REORDER_SELF_SWAP_4_STEP_2(SRC,TMP,SAMPLES,CHNUM,S0,S1,S2,S3) \
-for (i = 0; i < SAMPLES; i += CHNUM) {\
-    TMP = SRC[i+S0];\
-    SRC[i+S0] = SRC[i+S2];\
-    SRC[i+S2] = TMP;\
-    TMP = SRC[i+S1];\
-    SRC[i+S1] = SRC[i+S3];\
-    SRC[i+S3] = TMP;\
-}
-
-static int reorder_self_4_step_2(void *src, unsigned int samples,
-                                 unsigned int samplesize, unsigned int chnum,
-                                 int s0, int s1, int s2, int s3)
-{
-    int i;
-    switch (samplesize) {
-    case 3:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp0, tmp1, tmp2;
-        for (i = 0; i < samples * 3; i += chnum * 3) {
-            tmp0 = src_8[i+s0*3];
-            tmp1 = src_8[i+s0*3+1];
-            tmp2 = src_8[i+s0*3+2];
-            src_8[i+s0*3]   = src_8[i+s2*3];
-            src_8[i+s0*3+1] = src_8[i+s2*3+1];
-            src_8[i+s0*3+2] = src_8[i+s2*3+2];
-            src_8[i+s2*3]   = tmp0;
-            src_8[i+s2*3+1] = tmp1;
-            src_8[i+s2*3+2] = tmp2;
-            tmp0 = src_8[i+s1*3];
-            tmp1 = src_8[i+s1*3+1];
-            tmp2 = src_8[i+s1*3+2];
-            src_8[i+s1*3]   = src_8[i+s3*3];
-            src_8[i+s1*3+1] = src_8[i+s3*3+1];
-            src_8[i+s1*3+2] = src_8[i+s3*3+2];
-            src_8[i+s3*3]   = tmp0;
-            src_8[i+s3*3+1] = tmp1;
-            src_8[i+s3*3+2] = tmp2;
-        }
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-#define REORDER_SELF_SWAP_5_STEP_1(SRC,TMP,SAMPLES,CHNUM,S0,S1,S2,S3,S4) \
-for (i = 0; i < SAMPLES; i += CHNUM) {\
-    TMP = SRC[i+S0];\
-    SRC[i+S0] = SRC[i+S1];\
-    SRC[i+S1] = SRC[i+S2];\
-    SRC[i+S2] = SRC[i+S3];\
-    SRC[i+S3] = SRC[i+S4];\
-    SRC[i+S4] = TMP;\
-}
-
-static int reorder_self_5_step_1(void *src, unsigned int samples,
-                                 unsigned int samplesize, unsigned int chnum,
-                                 int s0, int s1, int s2, int s3, int s4)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_5_STEP_1(src_8,tmp,samples,6,s0,s1,s2,s3,s4);
-        }
-        else if (chnum==8) {
-            REORDER_SELF_SWAP_5_STEP_1(src_8,tmp,samples,8,s0,s1,s2,s3,s4);
-        }
-        else {
-            REORDER_SELF_SWAP_5_STEP_1(src_8,tmp,samples,5,s0,s1,s2,s3,s4);
-        }
-        break;
-    }
-    case 2:
-    {
-        int16_t *src_16 = src;
-        int16_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_5_STEP_1(src_16,tmp,samples,6,s0,s1,s2,s3,s4);
-        }
-        else if (chnum==8) {
-            REORDER_SELF_SWAP_5_STEP_1(src_16,tmp,samples,8,s0,s1,s2,s3,s4);
-        }
-        else {
-            REORDER_SELF_SWAP_5_STEP_1(src_16,tmp,samples,5,s0,s1,s2,s3,s4);
-        }
-        break;
-    }
-    case 3:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp0, tmp1, tmp2;
-        for (i = 0; i < samples * 3; i += chnum * 3) {
-            tmp0 = src_8[i+s0*3];
-            tmp1 = src_8[i+s0*3+1];
-            tmp2 = src_8[i+s0*3+2];
-            src_8[i+s0*3]   = src_8[i+s1*3];
-            src_8[i+s0*3+1] = src_8[i+s1*3+1];
-            src_8[i+s0*3+2] = src_8[i+s1*3+2];
-            src_8[i+s1*3]   = src_8[i+s2*3];
-            src_8[i+s1*3+1] = src_8[i+s2*3+1];
-            src_8[i+s1*3+2] = src_8[i+s2*3+2];
-            src_8[i+s2*3]   = src_8[i+s3*3];
-            src_8[i+s2*3+1] = src_8[i+s3*3+1];
-            src_8[i+s2*3+2] = src_8[i+s3*3+2];
-            src_8[i+s3*3]   = src_8[i+s4*3];
-            src_8[i+s3*3+1] = src_8[i+s4*3+1];
-            src_8[i+s3*3+2] = src_8[i+s4*3+2];
-            src_8[i+s4*3]   = tmp0;
-            src_8[i+s4*3+1] = tmp1;
-            src_8[i+s4*3+2] = tmp2;
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *src_32 = src;
-        int32_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_5_STEP_1(src_32,tmp,samples,6,s0,s1,s2,s3,s4);
-        }
-        else if (chnum==8) {
-            REORDER_SELF_SWAP_5_STEP_1(src_32,tmp,samples,8,s0,s1,s2,s3,s4);
-        }
-        else {
-            REORDER_SELF_SWAP_5_STEP_1(src_32,tmp,samples,5,s0,s1,s2,s3,s4);
-        }
-        break;
-    }
-    case 8:
-    {
-        int64_t *src_64 = src;
-        int64_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_5_STEP_1(src_64,tmp,samples,6,s0,s1,s2,s3,s4);
-        }
-        else if (chnum==8) {
-            REORDER_SELF_SWAP_5_STEP_1(src_64,tmp,samples,8,s0,s1,s2,s3,s4);
-        }
-        else {
-            REORDER_SELF_SWAP_5_STEP_1(src_64,tmp,samples,5,s0,s1,s2,s3,s4);
-        }
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-#define REORDER_SELF_SWAP_2_3(SRC,TMP,SAMPLES,CHNUM,S0,S1,S2,S3,S4) \
-for (i = 0; i < SAMPLES; i += CHNUM) {\
-    TMP = SRC[i+S0];\
-    SRC[i+S0] = SRC[i+S1];\
-    SRC[i+S1] = TMP;\
-    TMP = SRC[i+S2];\
-    SRC[i+S2] = SRC[i+S3];\
-    SRC[i+S3] = SRC[i+S4];\
-    SRC[i+S4] = TMP;\
-}
-
-static int reorder_self_2_3(void *src, unsigned int samples,
-                            unsigned int samplesize,
-                            int s0, int s1, int s2, int s3, int s4)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp;
-        REORDER_SELF_SWAP_2_3(src_8,tmp,samples,6,s0,s1,s2,s3,s4);
-        break;
-    }
-    case 2:
-    {
-        int16_t *src_16 = src;
-        int16_t tmp;
-        REORDER_SELF_SWAP_2_3(src_16,tmp,samples,6,s0,s1,s2,s3,s4);
-        break;
-    }
-    case 3:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp0, tmp1, tmp2;
-        for (i = 0; i < samples * 3; i += 18) {
-            tmp0 = src_8[i+s0*3];
-            tmp1 = src_8[i+s0*3+1];
-            tmp2 = src_8[i+s0*3+2];
-            src_8[i+s0*3]   = src_8[i+s1*3];
-            src_8[i+s0*3+1] = src_8[i+s1*3+1];
-            src_8[i+s0*3+2] = src_8[i+s1*3+2];
-            src_8[i+s1*3]   = tmp0;
-            src_8[i+s1*3+1] = tmp1;
-            src_8[i+s1*3+2] = tmp2;
-            tmp0 = src_8[i+s2*3];
-            tmp1 = src_8[i+s2*3+1];
-            tmp2 = src_8[i+s2*3+2];
-            src_8[i+s2*3]   = src_8[i+s3*3];
-            src_8[i+s2*3+1] = src_8[i+s3*3+1];
-            src_8[i+s2*3+2] = src_8[i+s3*3+2];
-            src_8[i+s3*3]   = src_8[i+s4*3];
-            src_8[i+s3*3+1] = src_8[i+s4*3+1];
-            src_8[i+s3*3+2] = src_8[i+s4*3+2];
-            src_8[i+s4*3]   = tmp0;
-            src_8[i+s4*3+1] = tmp1;
-            src_8[i+s4*3+2] = tmp2;
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *src_32 = src;
-        int32_t tmp;
-        REORDER_SELF_SWAP_2_3(src_32,tmp,samples,6,s0,s1,s2,s3,s4);
-        break;
-    }
-    case 8:
-    {
-        int64_t *src_64 = src;
-        int64_t tmp;
-        REORDER_SELF_SWAP_2_3(src_64,tmp,samples,6,s0,s1,s2,s3,s4);
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-#define REORDER_SELF_SWAP_3_3(SRC,TMP,SAMPLES,CHNUM,S0,S1,S2,S3,S4,S5) \
-for (i = 0; i < SAMPLES; i += CHNUM) {\
-    TMP = SRC[i+S0];\
-    SRC[i+S0] = SRC[i+S1];\
-    SRC[i+S1] = SRC[i+S2];\
-    SRC[i+S2] = TMP;\
-    TMP = SRC[i+S3];\
-    SRC[i+S3] = SRC[i+S4];\
-    SRC[i+S4] = SRC[i+S5];\
-    SRC[i+S5] = TMP;\
-}
-
-static int reorder_self_3_3(void *src, unsigned int samples,
-                            unsigned int samplesize,
-                            int s0, int s1, int s2, int s3, int s4, int s5)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp;
-        REORDER_SELF_SWAP_3_3(src_8,tmp,samples,6,s0,s1,s2,s3,s4,s5);
-        break;
-    }
-    case 2:
-    {
-        int16_t *src_16 = src;
-        int16_t tmp;
-        REORDER_SELF_SWAP_3_3(src_16,tmp,samples,6,s0,s1,s2,s3,s4,s5);
-        break;
-    }
-    case 3:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp0, tmp1, tmp2;
-        for (i = 0; i < samples * 3; i += 18) {
-            tmp0 = src_8[i+s0*3];
-            tmp1 = src_8[i+s0*3+1];
-            tmp2 = src_8[i+s0*3+2];
-            src_8[i+s0*3]   = src_8[i+s1*3];
-            src_8[i+s0*3+1] = src_8[i+s1*3+1];
-            src_8[i+s0*3+2] = src_8[i+s1*3+2];
-            src_8[i+s1*3]   = src_8[i+s2*3];
-            src_8[i+s1*3+1] = src_8[i+s2*3+1];
-            src_8[i+s1*3+2] = src_8[i+s2*3+2];
-            src_8[i+s2*3]   = tmp0;
-            src_8[i+s2*3+1] = tmp1;
-            src_8[i+s2*3+2] = tmp2;
-            tmp0 = src_8[i+s3*3];
-            tmp1 = src_8[i+s3*3+1];
-            tmp2 = src_8[i+s3*3+2];
-            src_8[i+s3*3]   = src_8[i+s4*3];
-            src_8[i+s3*3+1] = src_8[i+s4*3+1];
-            src_8[i+s3*3+2] = src_8[i+s4*3+2];
-            src_8[i+s4*3]   = src_8[i+s5*3];
-            src_8[i+s4*3+1] = src_8[i+s5*3+1];
-            src_8[i+s4*3+2] = src_8[i+s5*3+2];
-            src_8[i+s5*3]   = tmp0;
-            src_8[i+s5*3+1] = tmp1;
-            src_8[i+s5*3+2] = tmp2;
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *src_32 = src;
-        int32_t tmp;
-        REORDER_SELF_SWAP_3_3(src_32,tmp,samples,6,s0,s1,s2,s3,s4,s5);
-        break;
-    }
-    case 8:
-    {
-        int64_t *src_64 = src;
-        int64_t tmp;
-        REORDER_SELF_SWAP_3_3(src_64,tmp,samples,6,s0,s1,s2,s3,s4,s5);
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-#define REORDER_SELF_SWAP_2_4(SRC,TMP,SAMPLES,CHNUM,S0,S1,S2,S3,S4,S5) \
-for (i = 0; i < SAMPLES; i += CHNUM) {\
-    TMP = SRC[i+S0];\
-    SRC[i+S0] = SRC[i+S1];\
-    SRC[i+S1] = TMP;\
-    TMP = SRC[i+S2];\
-    SRC[i+S2] = SRC[i+S3];\
-    SRC[i+S3] = SRC[i+S4];\
-    SRC[i+S4] = SRC[i+S5];\
-    SRC[i+S5] = TMP;\
-}
-
-static int reorder_self_2_4(void *src, unsigned int samples,
-                            unsigned int samplesize, int chnum,
-                            int s0, int s1, int s2, int s3, int s4, int s5)
-{
-    int i;
-    switch (samplesize) {
-    case 1:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_2_4(src_8,tmp,samples,6,s0,s1,s2,s3,s4,s5);
-        } else {
-            REORDER_SELF_SWAP_2_4(src_8,tmp,samples,8,s0,s1,s2,s3,s4,s5);
-        }
-        break;
-    }
-    case 2:
-    {
-        int16_t *src_16 = src;
-        int16_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_2_4(src_16,tmp,samples,6,s0,s1,s2,s3,s4,s5);
-        } else {
-            REORDER_SELF_SWAP_2_4(src_16,tmp,samples,8,s0,s1,s2,s3,s4,s5);
-        }
-        break;
-    }
-    case 3:
-    {
-        int8_t *src_8 = src;
-        int8_t tmp0, tmp1, tmp2;
-        for (i = 0; i < samples * 3; i += chnum * 3) {
-            tmp0 = src_8[i+s0*3];
-            tmp1 = src_8[i+s0*3+1];
-            tmp2 = src_8[i+s0*3+2];
-            src_8[i+s0*3]   = src_8[i+s1*3];
-            src_8[i+s0*3+1] = src_8[i+s1*3+1];
-            src_8[i+s0*3+2] = src_8[i+s1*3+2];
-            src_8[i+s1*3]   = tmp0;
-            src_8[i+s1*3+1] = tmp1;
-            src_8[i+s1*3+2] = tmp2;
-            tmp0 = src_8[i+s2*3];
-            tmp1 = src_8[i+s2*3+1];
-            tmp2 = src_8[i+s2*3+2];
-            src_8[i+s2*3]   = src_8[i+s3*3];
-            src_8[i+s2*3+1] = src_8[i+s3*3+1];
-            src_8[i+s2*3+2] = src_8[i+s3*3+2];
-            src_8[i+s3*3]   = src_8[i+s4*3];
-            src_8[i+s3*3+1] = src_8[i+s4*3+1];
-            src_8[i+s3*3+2] = src_8[i+s4*3+2];
-            src_8[i+s4*3]   = src_8[i+s5*3];
-            src_8[i+s4*3+1] = src_8[i+s5*3+1];
-            src_8[i+s4*3+2] = src_8[i+s5*3+2];
-            src_8[i+s5*3]   = tmp0;
-            src_8[i+s5*3+1] = tmp1;
-            src_8[i+s5*3+2] = tmp2;
-        }
-        break;
-    }
-    case 4:
-    {
-        int32_t *src_32 = src;
-        int32_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_2_4(src_32,tmp,samples,6,s0,s1,s2,s3,s4,s5);
-        } else {
-            REORDER_SELF_SWAP_2_4(src_32,tmp,samples,8,s0,s1,s2,s3,s4,s5);
-        }
-        break;
-    }
-    case 8:
-    {
-        int64_t *src_64 = src;
-        int64_t tmp;
-        if (chnum==6) {
-            REORDER_SELF_SWAP_2_4(src_64,tmp,samples,6,s0,s1,s2,s3,s4,s5);
-        } else {
-            REORDER_SELF_SWAP_2_4(src_64,tmp,samples,8,s0,s1,s2,s3,s4,s5);
-        }
-        break;
-    }
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_ch] Unsupported sample size: %d, please "
-               "report this error on the MPlayer mailing list.\n",samplesize);
-        return 0;
-    }
-    return 1;
-}
-
-void reorder_channel(void *src,
-                     int src_layout,
-                     int dest_layout,
-                     int samples,
-                     int samplesize)
-{
-    if (dest_layout==src_layout)
-        return;
-    if (!AF_IS_SAME_CH_NUM(dest_layout,src_layout)) {
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_channel] different channel count "
-               "between current and target: %x, %x\n",
-               AF_GET_CH_NUM_WITH_LFE(src_layout),
-               AF_GET_CH_NUM_WITH_LFE(dest_layout));
-        return;
-    }
-    switch ((src_layout<<16)|dest_layout) {
-    // AF_CHANNEL_LAYOUT_5_0_A   L R C Ls Rs
-    // AF_CHANNEL_LAYOUT_5_0_B   L R Ls Rs C
-    // AF_CHANNEL_LAYOUT_5_0_C   L C R Ls Rs
-    // AF_CHANNEL_LAYOUT_5_0_D   C L R Ls Rs
-    case AF_CHANNEL_LAYOUT_5_0_A << 16 | AF_CHANNEL_LAYOUT_5_0_B:
-        reorder_self_3(src, samples, samplesize, 5, 2, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_A << 16 | AF_CHANNEL_LAYOUT_5_0_C:
-        reorder_self_2(src, samples, samplesize, 5, 1, 2);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_A << 16 | AF_CHANNEL_LAYOUT_5_0_D:
-        reorder_self_3(src, samples, samplesize, 5, 2, 1, 0);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_B << 16 | AF_CHANNEL_LAYOUT_5_0_A:
-        reorder_self_3(src, samples, samplesize, 5, 4, 3, 2);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_B << 16 | AF_CHANNEL_LAYOUT_5_0_C:
-        reorder_self_4_step_1(src, samples, samplesize, 5, 4, 3, 2, 1);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_B << 16 | AF_CHANNEL_LAYOUT_5_0_D:
-        reorder_self_5_step_1(src, samples, samplesize, 5, 4, 3, 2, 1, 0);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_C << 16 | AF_CHANNEL_LAYOUT_5_0_A:
-        reorder_self_2(src, samples, samplesize, 5, 1, 2);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_C << 16 | AF_CHANNEL_LAYOUT_5_0_B:
-        reorder_self_4_step_1(src, samples, samplesize, 5, 1, 2, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_C << 16 | AF_CHANNEL_LAYOUT_5_0_D:
-        reorder_self_2(src, samples, samplesize, 5, 0, 1);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_D << 16 | AF_CHANNEL_LAYOUT_5_0_A:
-        reorder_self_3(src, samples, samplesize, 5, 0, 1, 2);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_D << 16 | AF_CHANNEL_LAYOUT_5_0_B:
-        reorder_self_5_step_1(src, samples, samplesize, 5, 0, 1, 2, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_0_D << 16 | AF_CHANNEL_LAYOUT_5_0_C:
-        reorder_self_2(src, samples, samplesize, 5, 0, 1);
-        break;
-    // AF_CHANNEL_LAYOUT_5_1_A   L R C LFE Ls Rs
-    // AF_CHANNEL_LAYOUT_5_1_B   L R Ls Rs C LFE
-    // AF_CHANNEL_LAYOUT_5_1_C   L C R Ls Rs LFE
-    // AF_CHANNEL_LAYOUT_5_1_D   C L R Ls Rs LFE
-    // AF_CHANNEL_LAYOUT_5_1_E   LFE L C R Ls Rs
-    case AF_CHANNEL_LAYOUT_5_1_A << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        if (samplesize != 3)
-            reorder_self_2(src, samples/2, samplesize*2, 3, 1, 2);
-        else
-            reorder_self_4_step_2(src, samples, samplesize, 6, 2, 3, 4, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_A << 16 | AF_CHANNEL_LAYOUT_5_1_C:
-        reorder_self_2_3(src, samples, samplesize, 1, 2, 3, 4, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_A << 16 | AF_CHANNEL_LAYOUT_5_1_D:
-        reorder_self_3_3(src, samples, samplesize, 2, 1, 0, 3, 4, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_B << 16 | AF_CHANNEL_LAYOUT_5_1_A:
-        if (samplesize != 3)
-            reorder_self_2(src, samples/2, samplesize*2, 3, 1, 2);
-        else
-            reorder_self_4_step_2(src, samples, samplesize, 6, 2, 3, 4, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_B << 16 | AF_CHANNEL_LAYOUT_5_1_C:
-        reorder_self_4_step_1(src, samples, samplesize, 6, 4, 3, 2, 1);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_B << 16 | AF_CHANNEL_LAYOUT_5_1_D:
-        reorder_self_5_step_1(src, samples, samplesize, 6, 4, 3, 2, 1, 0);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_B << 16 | AF_CHANNEL_LAYOUT_5_1_E:
-        reorder_self_2_4(src, samples, samplesize, 6, 2, 4, 5, 3, 1, 0);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_C << 16 | AF_CHANNEL_LAYOUT_5_1_A:
-        reorder_self_2_3(src, samples, samplesize, 1, 2, 5, 4, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_C << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        reorder_self_4_step_1(src, samples, samplesize, 6, 1, 2, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_C << 16 | AF_CHANNEL_LAYOUT_5_1_D:
-        reorder_self_2(src, samples, samplesize, 6, 0, 1);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_D << 16 | AF_CHANNEL_LAYOUT_5_1_A:
-        reorder_self_3_3(src, samples, samplesize, 0, 1, 2, 5, 4, 3);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_D << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        reorder_self_5_step_1(src, samples, samplesize, 6, 0, 1, 2, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_D << 16 | AF_CHANNEL_LAYOUT_5_1_C:
-        reorder_self_2(src, samples, samplesize, 6, 0, 1);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_E << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        reorder_self_2_4(src, samples, samplesize, 6, 2, 4, 0, 1, 3, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_5_1_F << 16 | AF_CHANNEL_LAYOUT_5_1_B:
-        reorder_self_2_4(src, samples, samplesize, 6, 3, 5, 0, 1, 2, 4);
-        break;
-    // AF_CHANNEL_LAYOUT_7_1_A   L R C LFE Ls Rs Rls Rrs
-    // AF_CHANNEL_LAYOUT_7_1_B   L R Ls Rs C LFE Rls Rrs
-    // AF_CHANNEL_LAYOUT_7_1_C   L C R Ls Rs LFE Rls Rrs
-    // AF_CHANNEL_LAYOUT_7_1_D   C L R Ls Rs Rls Rrs LFE
-    // AF_CHANNEL_LAYOUT_7_1_F   C L R LFE Ls Rs Rls Rrs
-    case AF_CHANNEL_LAYOUT_7_1_A << 16 | AF_CHANNEL_LAYOUT_7_1_B:
-    case AF_CHANNEL_LAYOUT_7_1_B << 16 | AF_CHANNEL_LAYOUT_7_1_A:
-        if (samplesize != 3)
-            reorder_self_2(src, samples/2, samplesize*2, 4, 1, 2);
-        else
-            reorder_self_4_step_2(src, samples, samplesize, 8, 2, 3, 4, 5);
-        break;
-    case AF_CHANNEL_LAYOUT_7_1_B << 16 | AF_CHANNEL_LAYOUT_7_1_D:
-        // First convert to AF_CHANNEL_LAYOUT_7_1_F
-        reorder_self_2_4(src, samples, samplesize, 8, 3, 5, 4, 2, 1, 0);
-        // then convert to AF_CHANNEL_LAYOUT_7_1_D
-        reorder_self_5_step_1(src, samples, samplesize, 8, 3, 4, 5, 6, 7);
-        break;
-    case AF_CHANNEL_LAYOUT_7_1_C << 16 | AF_CHANNEL_LAYOUT_7_1_B:
-        reorder_self_4_step_1(src, samples, samplesize, 8, 1, 2, 3, 4);
-        break;
-    case AF_CHANNEL_LAYOUT_7_1_F << 16 | AF_CHANNEL_LAYOUT_7_1_B:
-        reorder_self_2_4(src, samples, samplesize, 8, 3, 5, 0, 1, 2, 4);
-        break;
-    default:
-        mp_msg(MSGT_GLOBAL, MSGL_WARN,
-               "[reorder_channel] unsupported from %x to %x, %d * %d\n",
-               src_layout, dest_layout, samples, samplesize);
-    }
-}
-
-
-static int channel_layout_mapping_5ch[AF_CHANNEL_LAYOUT_SOURCE_NUM] = {
-    AF_CHANNEL_LAYOUT_ALSA_5CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_AAC_5CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_WAVEEX_5CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_LAVC_5CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_VORBIS_5CH_DEFAULT,
-};
-
-static int channel_layout_mapping_6ch[AF_CHANNEL_LAYOUT_SOURCE_NUM] = {
-    AF_CHANNEL_LAYOUT_ALSA_6CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_AAC_6CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_WAVEEX_6CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_LAVC_6CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_VORBIS_6CH_DEFAULT,
-};
-
-static int channel_layout_mapping_8ch[AF_CHANNEL_LAYOUT_SOURCE_NUM] = {
-    AF_CHANNEL_LAYOUT_ALSA_8CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_AAC_8CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_WAVEEX_8CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_LAVC_8CH_DEFAULT,
-    AF_CHANNEL_LAYOUT_VORBIS_8CH_DEFAULT,
-};
-
-void reorder_channel_copy_nch(void *src,
-                              int src_layout,
-                              void *dest,
-                              int dest_layout,
-                              int chnum,
-                              int samples,
-                              int samplesize)
-{
-    if (chnum < 5 || chnum == 7 || chnum > 8 ||
-            src_layout < 0 || dest_layout < 0 ||
-            src_layout >= AF_CHANNEL_LAYOUT_SOURCE_NUM ||
-            dest_layout >= AF_CHANNEL_LAYOUT_SOURCE_NUM)
-        memcpy(dest, src, samples*samplesize);
-    else if (chnum == 6)
-        reorder_channel_copy(src, channel_layout_mapping_6ch[src_layout],
-                             dest, channel_layout_mapping_6ch[dest_layout],
-                             samples, samplesize);
-    else if (chnum == 8)
-        reorder_channel_copy(src, channel_layout_mapping_8ch[src_layout],
-                             dest, channel_layout_mapping_8ch[dest_layout],
-                             samples, samplesize);
-    else
-        reorder_channel_copy(src, channel_layout_mapping_5ch[src_layout],
-                             dest, channel_layout_mapping_5ch[dest_layout],
-                             samples, samplesize);
-}
-
-void reorder_channel_nch(void *buf,
-                         int src_layout,
-                         int dest_layout,
-                         int chnum,
-                         int samples,
-                         int samplesize)
-{
-    if (src_layout == dest_layout || chnum < 5 || chnum == 7 || chnum > 8 ||
-            src_layout < 0 || dest_layout < 0 ||
-            src_layout >= AF_CHANNEL_LAYOUT_SOURCE_NUM ||
-            dest_layout >= AF_CHANNEL_LAYOUT_SOURCE_NUM ||
-            src_layout == dest_layout)
-        return;
-    if (chnum == 6)
-        reorder_channel(buf, channel_layout_mapping_6ch[src_layout],
-                        channel_layout_mapping_6ch[dest_layout],
-                        samples, samplesize);
-    else if (chnum == 8)
-        reorder_channel(buf, channel_layout_mapping_8ch[src_layout],
-                        channel_layout_mapping_8ch[dest_layout],
-                        samples, samplesize);
-    else
-        reorder_channel(buf, channel_layout_mapping_5ch[src_layout],
-                        channel_layout_mapping_5ch[dest_layout],
-                        samples, samplesize);
-}
-
-#ifdef TEST
-
-static void test_copy(int channels) {
-    int samples = 12*1024*1024;
-    int samplesize = 2;
-    int i;
-    unsigned char *bufin = malloc((samples+100)*samplesize);
-    unsigned char *bufout = malloc((samples+100)*samplesize);
-    memset(bufin, 0xFF, samples*samplesize);
-    for (i = 0;i < 100; ++i)
-        reorder_channel_copy(bufin, AF_CHANNEL_LAYOUT_5_1_A,
-                             bufout, AF_CHANNEL_LAYOUT_5_1_B,
-                             samples, samplesize);
-//    reorder_channel(bufin, AF_CHANNEL_LAYOUT_5_1_B,
-//                         AF_CHANNEL_LAYOUT_5_1_D,
-//                         samples, samplesize);
-    free(bufin);
-    free(bufout);
-}
-
-int main(int argc, char *argv[]) {
-    int channels = 6;
-    if (argc > 1)
-        channels = atoi(argv[1]);
-    test_copy(channels);
-    return 0;
-}
-
-#endif
-
 static inline void reorder_to_planar_(void *restrict out, const void *restrict in,
                                       size_t size, size_t nchan, size_t nmemb)
 {
diff --git a/audio/reorder_ch.h b/audio/reorder_ch.h
index 07daa35a08..c9c101e719 100644
--- a/audio/reorder_ch.h
+++ b/audio/reorder_ch.h
@@ -25,114 +25,6 @@
 
 #include <inttypes.h>
 
-// L   - Left
-// R   - Right
-// C   - Center
-// Ls  - Left Surround
-// Rs  - Right Surround
-// Cs  - Center Surround
-// Rls - Rear Left Surround
-// Rrs - Rear Right Surround
-
-#define AF_LFE   (1<<7)
-
-#define AF_CHANNEL_LAYOUT_MONO   ((100<<8)|1)
-#define AF_CHANNEL_LAYOUT_STEREO ((101<<8)|2)
-#define AF_CHANNEL_LAYOUT_1_0 AF_CHANNEL_LAYOUT_MONO   // C
-#define AF_CHANNEL_LAYOUT_2_0 AF_CHANNEL_LAYOUT_STEREO // L R
-#define AF_CHANNEL_LAYOUT_2_1   ((102<<8)|3)           // L R LFE
-#define AF_CHANNEL_LAYOUT_3_0_A ((103<<8)|3)           // L R C
-#define AF_CHANNEL_LAYOUT_3_0_B ((104<<8)|3)           // C L R
-#define AF_CHANNEL_LAYOUT_4_0_A ((105<<8)|4)           // L R C Cs
-#define AF_CHANNEL_LAYOUT_4_0_B ((106<<8)|4)           // C L R Cs
-#define AF_CHANNEL_LAYOUT_4_0_C ((107<<8)|4)           // L R Ls Rs
-#define AF_CHANNEL_LAYOUT_5_0_A ((108<<8)|5)           // L R C Ls Rs
-#define AF_CHANNEL_LAYOUT_5_0_B ((109<<8)|5)           // L R Ls Rs C
-#define AF_CHANNEL_LAYOUT_5_0_C ((110<<8)|5)           // L C R Ls Rs
-#define AF_CHANNEL_LAYOUT_5_0_D ((111<<8)|5)           // C L R Ls Rs
-#define AF_CHANNEL_LAYOUT_5_1_A ((112<<8)|6|AF_LFE)    // L R C LFE Ls Rs
-#define AF_CHANNEL_LAYOUT_5_1_B ((113<<8)|6|AF_LFE)    // L R Ls Rs C LFE
-#define AF_CHANNEL_LAYOUT_5_1_C ((114<<8)|6|AF_LFE)    // L C R Ls Rs LFE
-#define AF_CHANNEL_LAYOUT_5_1_D ((115<<8)|6|AF_LFE)    // C L R Ls Rs LFE
-#define AF_CHANNEL_LAYOUT_5_1_E ((116<<8)|6|AF_LFE)    // LFE L C R Ls Rs
-#define AF_CHANNEL_LAYOUT_5_1_F ((117<<8)|6|AF_LFE)    // C L R LFE Ls Rs
-#define AF_CHANNEL_LAYOUT_6_1_A ((118<<8)|7|AF_LFE)    // L R C LFE Ls Rs Cs
-#define AF_CHANNEL_LAYOUT_7_1_A ((119<<8)|8|AF_LFE)    // L R C LFE Ls Rs Rls Rrs
-#define AF_CHANNEL_LAYOUT_7_1_B ((120<<8)|8|AF_LFE)    // L R Ls Rs C LFE Rls Rrs
-#define AF_CHANNEL_LAYOUT_7_1_C ((121<<8)|8|AF_LFE)    // L C R Ls Rs LFE Rls Rrs
-#define AF_CHANNEL_LAYOUT_7_1_D ((122<<8)|8|AF_LFE)    // C L R Ls Rs Rls Rrs LFE
-#define AF_CHANNEL_LAYOUT_7_1_F ((123<<8)|8|AF_LFE)    // C L R LFE Ls Rs Rls Rrs
-
-
-#define AF_CHANNEL_LAYOUT_ALSA_5CH_DEFAULT AF_CHANNEL_LAYOUT_5_0_B
-#define AF_CHANNEL_LAYOUT_ALSA_6CH_DEFAULT AF_CHANNEL_LAYOUT_5_1_B
-#define AF_CHANNEL_LAYOUT_ALSA_8CH_DEFAULT AF_CHANNEL_LAYOUT_7_1_B
-#define AF_CHANNEL_LAYOUT_MPLAYER_5CH_DEFAULT AF_CHANNEL_LAYOUT_ALSA_5CH_DEFAULT
-#define AF_CHANNEL_LAYOUT_MPLAYER_6CH_DEFAULT AF_CHANNEL_LAYOUT_ALSA_6CH_DEFAULT
-#define AF_CHANNEL_LAYOUT_MPLAYER_8CH_DEFAULT AF_CHANNEL_LAYOUT_ALSA_8CH_DEFAULT
-#define AF_CHANNEL_LAYOUT_AAC_5CH_DEFAULT AF_CHANNEL_LAYOUT_5_0_D
-#define AF_CHANNEL_LAYOUT_AAC_6CH_DEFAULT AF_CHANNEL_LAYOUT_5_1_D
-#define AF_CHANNEL_LAYOUT_AAC_8CH_DEFAULT AF_CHANNEL_LAYOUT_7_1_D
-#define AF_CHANNEL_LAYOUT_WAVEEX_5CH_DEFAULT AF_CHANNEL_LAYOUT_5_0_A
-#define AF_CHANNEL_LAYOUT_WAVEEX_6CH_DEFAULT AF_CHANNEL_LAYOUT_5_1_A
-#define AF_CHANNEL_LAYOUT_WAVEEX_8CH_DEFAULT AF_CHANNEL_LAYOUT_7_1_A
-#define AF_CHANNEL_LAYOUT_LAVC_5CH_DEFAULT AF_CHANNEL_LAYOUT_5_0_A
-#define AF_CHANNEL_LAYOUT_LAVC_6CH_DEFAULT AF_CHANNEL_LAYOUT_5_1_A
-#define AF_CHANNEL_LAYOUT_LAVC_8CH_DEFAULT AF_CHANNEL_LAYOUT_7_1_A
-#define AF_CHANNEL_LAYOUT_VORBIS_5CH_DEFAULT AF_CHANNEL_LAYOUT_5_0_C
-#define AF_CHANNEL_LAYOUT_VORBIS_6CH_DEFAULT AF_CHANNEL_LAYOUT_5_1_C
-#define AF_CHANNEL_LAYOUT_VORBIS_8CH_DEFAULT AF_CHANNEL_LAYOUT_7_1_C
-
-#define AF_CHANNEL_MASK  0xFF
-#define AF_GET_CH_NUM(A) ((A)&0x7F)
-#define AF_GET_CH_NUM_WITH_LFE(A) ((A)&0xFF)
-#define AF_IS_SAME_CH_NUM(A,B) (((A)&0xFF)==((B)&0xFF))
-#define AF_IS_LAYOUT_SPECIFIED(A) ((A)&0xFFFFF800)
-#define AF_IS_LAYOUT_UNSPECIFIED(A) (!AF_IS_LAYOUT_SPECIFIED(A))
-
-/// Optimized channel reorder between channel layouts with same channel number.
-void reorder_channel_copy(void *src,
-                          int src_layout,
-                          void *dest,
-                          int dest_layout,
-                          int samples,
-                          int samplesize);
-
-/// Same with reorder_channel_copy, but works on single buffer.
-void reorder_channel(void *buf,
-                     int src_layout,
-                     int dest_layout,
-                     int samples,
-                     int samplesize);
-
-// Channel layout definitions for different audio sources or targets
-// When specified channel number, they will be map to the specific layouts.
-#define AF_CHANNEL_LAYOUT_ALSA_DEFAULT         0
-#define AF_CHANNEL_LAYOUT_AAC_DEFAULT          1
-#define AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT       2
-#define AF_CHANNEL_LAYOUT_LAVC_DEFAULT         3
-#define AF_CHANNEL_LAYOUT_VORBIS_DEFAULT       4
-#define AF_CHANNEL_LAYOUT_SOURCE_NUM           5
-#define AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT AF_CHANNEL_LAYOUT_ALSA_DEFAULT
-
-/// Optimized channel reorder between different audio sources and targets.
-void reorder_channel_copy_nch(void *src,
-                              int src_layout,
-                              void *dest,
-                              int dest_layout,
-                              int chnum,
-                              int samples,
-                              int samplesize);
-
-/// Same with reorder_channel_copy_nch, but works on single buffer.
-void reorder_channel_nch(void *buf,
-                         int src_layout,
-                         int dest_layout,
-                         int chnum,
-                         int samples,
-                         int samplesize);
-
-/// Utility function for planar audio conversions
 void reorder_to_planar(void *restrict out, const void *restrict in,
                        size_t size, size_t nchan, size_t nmemb);
 void reorder_to_packed(uint8_t *out, uint8_t **in,